├── INVITE-recv180-CANCEL-or-BYE.xml
├── INVITE-test-session-refresher-uac.xml
├── dtmf
    └── playdtmf.xml
├── fax
    ├── sendfax-wjdtest20150505.pcap
    └── sendfax.xml
└── out-in
    ├── beep_1sec_50x160b.alaw
    ├── i3pcc-oinv-o3pcc-i603.xml
    ├── i3pcc-oinv-o3pcc-ibye.xml
    ├── iinv-o180-icancel.xml
    ├── iinv-o180-wait-o200-ibye.xml
    ├── iinv-o180-wait-o200-iinv-o200-obye.xml
    ├── iinv-o180-wait-o200-wait-obye.xml
    ├── iinv-o200-iinv-o200-ibye.xml
    ├── iinv-o200-iinv-o200-obye.xml
    ├── iinv-o200-obye.xml
    ├── iinv-o200-oinv-i200-obye--w-cli-and-sessupd.xml
    ├── iinv-o200-oinv-i200-oi3pcc-orefer-obye.xml
    ├── iinv-o200-oinv-i200-oinv-i200-wait-obye.xml
    ├── iinv-o302.xml
    ├── iinv-o486.xml
    ├── oinv-i180-ocancel.xml
    ├── oinv-i200-ibye--w-rpid.xml
    ├── oinv-i200-ibye--w-sesstimer-uac.xml
    ├── oinv-i200-ibye.xml
    ├── oinv-i200-iinv-o200-ibye.xml
    ├── oinv-i200-oinv-i200-oi3pcc-orefer-obye.xml
    ├── oinv-i486.xml
    └── oreg-i200.xml


/INVITE-recv180-CANCEL-or-BYE.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <!--
  4 | 
  5 | This SIPp scenario does an:
  6 |  - INVITE
  7 |  - waits for 180 or 183 (ringing)
  8 |  - waits some more
  9 |  - CANCELs the call
 10 | 
 11 | Parameters:
 12 |  -sf invite-and-cancel.xml 
 13 |  -m 1
 14 |  -s authuser
 15 |  -ap password
 16 |  -key fromuser fromuser
 17 |  -key domain domain
 18 |  -key dnid destination
 19 |  -key clid callerid-number
 20 |  -i local_ip
 21 |  hostname
 22 |  (set TERM=dumb to get at least a term)
 23 | 
 24 | Features:
 25 |  - All recv calls have a timeout of (default) 8 seconds. That should
 26 |    be enough even for crappy links.
 27 |  - After ringing, we wait a bit before calling CANCEL. If the UAS picks
 28 |    up in the mean time, we handle it gracefully.
 29 |  - Handled timeouts to not return an errorcode. Unhandled timeouts (and
 30 |    other errors) do.
 31 | 
 32 | Questions:
 33 |  - Do we need more codecs? Or less?
 34 | 
 35 | Author:
 36 |  Walter Doekes, april 2014.
 37 | 
 38 | -->
 39 | <scenario name="INVITE-recv180-CANCEL-or-BYE">
 40 | 
 41 |   <!-- ************************************************************* -->
 42 |   <!-- Start, send INVITE, resend with auth -->
 43 |   <!-- ************************************************************* -->
 44 | 
 45 |   <send retrans="500" start_txn="invite">
 46 |     <![CDATA[
 47 | 
 48 |       INVITE sip:[dnid]@[domain] SIP/2.0
 49 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 50 |       From: sip:[fromuser]@[domain];tag=[pid]SIPpTag00[call_number]
 51 |       To: sip:[dnid]@[domain]
 52 |       Contact: sip:[service]@[local_ip]:[local_port]
 53 |       Call-ID: [call_id]
 54 |       CSeq: [cseq] INVITE
 55 |       Max-Forwards: 30
 56 |       Content-Type: application/sdp
 57 |       Content-Length: [len]
 58 |       P-Asserted-Identity: "" <sip:[clid]@[domain]>
 59 |       Remote-Party-ID: "" <sip:[clid]@[domain]>;party=calling;privacy=off;screen=no
 60 | 
 61 |       v=0
 62 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 63 |       s=-
 64 |       c=IN IP[media_ip_type] [media_ip]
 65 |       t=0 0
 66 |       m=audio [media_port] RTP/AVP 8 0
 67 |       a=rtpmap:8 PCMA/8000
 68 |       a=rtpmap:0 PCMU/8000
 69 | 
 70 |     ]]>
 71 |   </send>
 72 | 
 73 |   <recv response="100" optional="true" response_txn="invite" timeout="8000"/>
 74 | 
 75 |   <recv response="401" optional="true" next="invite-with-auth" auth="true" response_txn="invite" timeout="8000">
 76 |     <action>
 77 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 78 |       <ereg regexp=";.*" search_in="hdr" header="To" check_it="false" assign_to="2"/>
 79 |     </action>
 80 |   </recv>
 81 | 
 82 |   <recv response="407" auth="true" response_txn="invite" timeout="8000">
 83 |     <action>
 84 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 85 |       <ereg regexp=";.*" search_in="hdr" header="To" check_it="false" assign_to="2"/>
 86 |     </action>
 87 |   </recv>
 88 | 
 89 |   <label id="invite-with-auth"/>
 90 | 
 91 |   <send ack_txn="invite">
 92 |     <![CDATA[
 93 | 
 94 |       ACK sip:[dnid]@[domain] SIP/2.0
 95 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$1]
 96 |       From: sip:[fromuser]@[domain];tag=[pid]SIPpTag00[call_number]
 97 |       To: sip:[dnid]@[domain][$2]
 98 |       Call-ID: [call_id]
 99 |       CSeq: [cseq] ACK
100 |       Max-Forwards: 30
101 |       Content-Length: 0
102 | 
103 |     ]]>
104 |   </send>
105 | 
106 |   <send retrans="500" start_txn="invite">
107 |     <![CDATA[
108 | 
109 |       INVITE sip:[dnid]@[domain] SIP/2.0
110 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
111 |       From: sip:[fromuser]@[domain];tag=[pid]SIPpTag00[call_number]
112 |       To: sip:[dnid]@[domain]
113 |       Contact: sip:[service]@[local_ip]:[local_port]
114 |       [authentication]
115 |       Call-ID: [call_id]
116 |       CSeq: [cseq] INVITE
117 |       Max-Forwards: 30
118 |       Content-Type: application/sdp
119 |       Content-Length: [len]
120 |       P-Asserted-Identity: "" <sip:[clid]@[domain]>
121 |       Remote-Party-ID: "" <sip:[clid]@[domain]>;party=calling;privacy=off;screen=no
122 | 
123 |       v=0
124 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
125 |       s=-
126 |       c=IN IP[media_ip_type] [media_ip]
127 |       t=0 0
128 |       m=audio [media_port] RTP/AVP 8 0
129 |       a=rtpmap:8 PCMA/8000
130 |       a=rtpmap:0 PCMU/8000
131 | 
132 |     ]]>
133 |   </send>
134 | 
135 |   <recv response="100" response_txn="invite" timeout="8000">
136 |     <action>
137 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
138 |     </action>
139 |   </recv>
140 | 
141 |   <!-- The 180 and 183 have to have a timeout, or else the timeout on
142 |        the 200 doesn't work. We also need next, or we'd get a wait on
143 |        the the 183 (if we only get a 180). -->
144 |   <recv response="180" optional="true" response_txn="invite" next="wait" timeout="10000" ontimeout="wait">
145 |     <action>
146 |       <assignstr assign_to="3" value="1"/>
147 |     </action>
148 |   </recv>
149 | 
150 |   <recv response="183" optional="true" response_txn="invite" next="wait" timeout="10000" ontimeout="wait">
151 |     <action>
152 |       <assignstr assign_to="3" value="1"/>
153 |     </action>
154 |   </recv>
155 | 
156 |   <label id="wait"/>
157 | 
158 |   <!-- Wait 1.5 seconds for a 200. If we got it, handle ACK+BYE. If we
159 |        didn't, continue on to cancel. -->
160 |   <recv response="200" rrs="true" response_txn="invite" next="ack-bye" timeout="1500" ontimeout="cancel">
161 |     <action>
162 |       <assignstr assign_to="3" value="1"/>
163 |       <ereg regexp=";.*" search_in="hdr" header="To" check_it="false" assign_to="1"/>
164 |     </action>
165 |   </recv>
166 | 
167 | 
168 |   <!-- ************************************************************* -->
169 |   <!-- Got 180/183, send CANCEL -->
170 |   <!-- ************************************************************* -->
171 | 
172 |   <label id="cancel"/>
173 | 
174 |   <send>
175 |     <![CDATA[
176 | 
177 |       CANCEL sip:[dnid]@[domain] SIP/2.0
178 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$1]
179 |       From: sip:[fromuser]@[domain];tag=[pid]SIPpTag00[call_number]
180 |       To: sip:[dnid]@[domain]
181 |       Call-ID: [call_id]
182 |       CSeq: [cseq] CANCEL
183 |       Max-Forwards: 30
184 | 
185 |     ]]>
186 |   </send>
187 |   
188 |   <recv response="487" response_txn="invite" timeout="8000">
189 |     <action>
190 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
191 |       <ereg regexp=";.*" search_in="hdr" header="To" check_it="false" assign_to="2"/>
192 |     </action>
193 |   </recv>
194 | 
195 |   <send ack_txn="invite">
196 |     <![CDATA[
197 | 
198 |       ACK sip:[dnid]@[domain] SIP/2.0
199 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$1]
200 |       From: sip:[fromuser]@[domain];tag=[pid]SIPpTag00[call_number]
201 |       To: sip:[dnid]@[domain][$2]
202 |       Call-ID: [call_id]
203 |       CSeq: [cseq] ACK
204 |       Max-Forwards: 30
205 |       Content-Length: 0
206 | 
207 |     ]]>
208 |   </send>
209 | 
210 |   <!-- Check that we got at least a 180, 183 or 200. -->
211 |   <nop hide="true">
212 |     <action>
213 |       <ereg regexp="1" search_in="var" variable="3" check_it="true" assign_to="4"/>
214 |     </action>
215 |   </nop>    
216 |   <nop hide="true" test="4" next="done"/>
217 | 
218 |   <!-- sipp did not return non-zero with check_it="true" above. But with
219 |        the timeout of 1 here, it does. -->
220 |   <recv response="999" timeout="1"/><!-- die die die! -->
221 | 
222 | 
223 |   <!-- ************************************************************* -->
224 |   <!-- Got 200, send ACK then BYE -->
225 |   <!-- ************************************************************* -->
226 | 
227 |   <label id="ack-bye"/>
228 | 
229 |   <send ack_txn="invite">
230 |     <![CDATA[
231 | 
232 |       ACK [next_url] SIP/2.0
233 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
234 |       [routes]
235 |       From: sip:[fromuser]@[domain];tag=[pid]SIPpTag00[call_number]
236 |       To: sip:[dnid]@[domain][$1]
237 |       Call-ID: [call_id]
238 |       CSeq: [cseq] ACK
239 |       Max-Forwards: 30
240 |       Content-Length: 0
241 | 
242 |     ]]>
243 |   </send>
244 | 
245 |   <send start_txn="bye">
246 |     <![CDATA[
247 | 
248 |       BYE [next_url] SIP/2.0
249 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
250 |       [routes]
251 |       From: sip:[fromuser]@[domain];tag=[pid]SIPpTag00[call_number]
252 |       To: sip:[dnid]@[domain][$1]
253 |       Call-ID: [call_id]
254 |       CSeq: [cseq] BYE
255 |       Max-Forwards: 30
256 |       Content-Length: 0
257 | 
258 |     ]]>
259 |   </send>
260 | 
261 |   <recv response="200" response_txn="bye" timeout="8000"/>
262 | 
263 | 
264 |   <!-- ************************************************************* -->
265 |   <!-- Done -->
266 |   <!-- ************************************************************* -->
267 | 
268 |   <label id="done"/>
269 | 
270 | </scenario>
271 | 


--------------------------------------------------------------------------------
/INVITE-test-session-refresher-uac.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="INVITE-test-session-refresher-uac">
  4 | 
  5 |   <!--
  6 | 
  7 |     Test UAS when we (UAC) do the refreshing for bug ASTERISK-22551,
  8 |     Walter Doekes, May 2014
  9 | 
 10 |     Asterisk dialplan:
 11 | 
 12 |       exten => wait_a_while,1,Ringing()
 13 |        ;; time is now 00:00
 14 |        same => n,Wait(30)
 15 |        ;; time is now 00:30
 16 |        same => n,Answer()
 17 |        ;; if time is 00:30 + 90/2, it's refresh time: 01:15 
 18 |        ;; however, asterisk thinks it's disconnect time
 19 |        ;; at: 00:00 + 90 - min(90/3, 32) = 00:58
 20 |        ;;
 21 |        ;; wait a long while.. let the sipp script do its testing.
 22 |        ;; within these 300 seconds we should send the BYE, but at
 23 |        ;; the right time.
 24 |        same => n,Wait(300)
 25 |        ;; if we get here, things are broken.
 26 |        same => n,Hangup()
 27 | 
 28 |   -->
 29 | 
 30 |   <send retrans="500" start_txn="invite">
 31 |     <![CDATA[
 32 | 
 33 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 34 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 35 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 36 |       To: sip:[tel]@[remote_ip]:[remote_port]
 37 |       Contact: sip:[service]@[local_ip]:[local_port]
 38 |       Call-ID: [call_id]
 39 |       CSeq: [cseq] INVITE
 40 |       Max-Forwards: 70
 41 |       Content-Type: application/sdp
 42 |       Content-Length: [len]
 43 |       Require: timer
 44 |       Session-Expires: 90;refresher=uac
 45 |       Supported: timer
 46 | 
 47 |       v=0
 48 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 49 |       s=-
 50 |       c=IN IP[media_ip_type] [media_ip]
 51 |       t=0 0
 52 |       m=audio [media_port] RTP/AVP 8 0
 53 |       a=rtpmap:8 PCMA/8000
 54 |       a=rtpmap:0 PCMU/8000
 55 | 
 56 |     ]]>
 57 |   </send>
 58 | 
 59 |   <recv response="100" optional="true" response_txn="invite"/>
 60 | 
 61 |   <recv response="401" optional="true" next="invite-with-auth" auth="true" response_txn="invite">
 62 |     <action>
 63 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 64 |     </action>
 65 |   </recv>
 66 | 
 67 |   <recv response="407" auth="true" response_txn="invite">
 68 |     <action>
 69 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 70 |     </action>
 71 |   </recv>
 72 | 
 73 |   <label id="invite-with-auth"/>
 74 | 
 75 |   <!-- ACK is within transaction because of non-2xx. Reuse Via branch.      -->
 76 |   <!-- We don't use last_Via: because it would include rport stuff from     -->
 77 |   <!-- remote.                                                              -->
 78 |   <send ack_txn="invite">
 79 |     <![CDATA[
 80 | 
 81 |       ACK sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 82 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$1]
 83 |       [last_From:]
 84 |       [last_To:]
 85 |       [last_Call-ID:]
 86 |       CSeq: [cseq] ACK
 87 |       Max-Forwards: 70
 88 |       Content-Length: 0
 89 | 
 90 |     ]]>
 91 |   </send>
 92 | 
 93 |   <send retrans="500" start_txn="invite">
 94 |     <![CDATA[
 95 | 
 96 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 97 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 98 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 99 |       To: sip:[tel]@[remote_ip]:[remote_port]
100 |       Contact: sip:[service]@[local_ip]:[local_port]
101 |       [authentication]
102 |       Call-ID: [call_id]
103 |       CSeq: [cseq] INVITE
104 |       Max-Forwards: 70
105 |       Content-Type: application/sdp
106 |       Content-Length: [len]
107 |       Require: timer
108 |       Session-Expires: 90;refresher=uac
109 |       Supported: timer
110 | 
111 |       v=0
112 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
113 |       s=-
114 |       c=IN IP[media_ip_type] [media_ip]
115 |       t=0 0
116 |       m=audio [media_port] RTP/AVP 8 0
117 |       a=rtpmap:8 PCMA/8000
118 |       a=rtpmap:0 PCMU/8000
119 | 
120 |     ]]>
121 |   </send>
122 | 
123 |   <recv response="100" response_txn="invite"/>
124 | 
125 |   <recv response="180" optional="true" response_txn="invite"/>
126 | 
127 |   <recv response="183" optional="true" response_txn="invite"/>
128 | 
129 |   <!--
130 | 
131 |     We expect the UAS to wait for 30 seconds in ringing state here.
132 |     This triggers the bug as explained below.
133 | 
134 |   -->
135 |   
136 |   <recv response="200" rrs="true" response_txn="invite"/>
137 | 
138 |   <send ack_txn="invite">
139 |     <![CDATA[
140 | 
141 |       ACK [next_url] SIP/2.0
142 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
143 |       [routes]
144 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
145 |       To: sip:[tel]@[remote_ip]:[remote_port][peer_tag_param]
146 |       Contact: sip:[service]@[local_ip]:[local_port]
147 |       Call-ID: [call_id]
148 |       CSeq: [cseq] ACK
149 |       Max-Forwards: 70
150 |       Content-Type: application/sdp
151 |       Content-Length: [len]
152 | 
153 |       v=0
154 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
155 |       s=-
156 |       c=IN IP[media_ip_type] [media_ip]
157 |       t=0 0
158 |       m=audio [media_port] RTP/AVP 8 0
159 |       a=rtpmap:8 PCMA/8000
160 |       a=rtpmap:0 PCMU/8000
161 |     ]]>
162 |   </send>
163 | 
164 |   <pause milliseconds="45000"/>
165 | 
166 |   <!--
167 | 
168 |     If we're still alive, we're at t75 (30 ringing + 45 pause).
169 | 
170 |     If bug ASTERISK-22551 is still in effect, Asterisk will have
171 |     counted from the start of the INVITE t0 to t58 and decided
172 |     that is was kill time.
173 | 
174 |     If it is fixed, we have time until t88 to send the refresh reINVITE.
175 |     Let's do that.
176 | 
177 |   -->
178 | 
179 |   <send retrans="500" start_txn="invite">
180 |     <![CDATA[
181 | 
182 |       INVITE [next_url] SIP/2.0
183 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
184 |       [routes]
185 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
186 |       To: sip:[tel]@[remote_ip]:[remote_port][peer_tag_param]
187 |       Contact: sip:[service]@[local_ip]:[local_port]
188 |       Call-ID: [call_id]
189 |       CSeq: [cseq] INVITE
190 |       Max-Forwards: 70
191 |       Content-Type: application/sdp
192 |       Content-Length: [len]
193 |       Require: timer
194 |       Session-Expires: 90;refresher=uac
195 |       Supported: timer
196 |       X-Reason: SIP re-invite (Session-Timers)
197 | 
198 |       v=0
199 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
200 |       s=-
201 |       c=IN IP[media_ip_type] [media_ip]
202 |       t=0 0
203 |       m=audio [media_port] RTP/AVP 8 0
204 |       a=rtpmap:8 PCMA/8000
205 |       a=rtpmap:0 PCMU/8000
206 | 
207 |     ]]>
208 |   </send>
209 | 
210 |   <recv response="200" response_txn="invite"/>
211 | 
212 |   <send ack_txn="invite">
213 |     <![CDATA[
214 | 
215 |       ACK [next_url] SIP/2.0
216 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
217 |       [routes]
218 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
219 |       To: sip:[tel]@[remote_ip]:[remote_port][peer_tag_param]
220 |       Contact: sip:[service]@[local_ip]:[local_port]
221 |       Call-ID: [call_id]
222 |       CSeq: [cseq] ACK
223 |       Max-Forwards: 70
224 |       Content-Type: application/sdp
225 |       Content-Length: [len]
226 | 
227 |       v=0
228 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
229 |       s=-
230 |       c=IN IP[media_ip_type] [media_ip]
231 |       t=0 0
232 |       m=audio [media_port] RTP/AVP 8 0
233 |       a=rtpmap:8 PCMA/8000
234 |       a=rtpmap:0 PCMU/8000
235 |     ]]>
236 |   </send>
237 | 
238 |   <!--
239 | 
240 |     For extra testing, wait another 50 seconds. We don't need to send
241 |     another refresh for 58 seconds.
242 | 
243 |   -->
244 | 
245 |   <pause milliseconds="50000"/>
246 | 
247 |   <!--
248 | 
249 |     In 8 seconds we should be receiving a BYE.
250 | 
251 |     Give asterisk 15 seconds before aborting this scenario with failure.
252 | 
253 |   -->
254 | 
255 |   <recv request="BYE" timeout="15000"/>
256 | 
257 |   <send>
258 |     <![CDATA[
259 | 
260 |       SIP/2.0 200 OK
261 |       [last_Via:]
262 |       [last_Record-Route:]
263 |       [last_From:]
264 |       [last_To:]
265 |       [last_Call-ID:]
266 |       [last_CSeq:]
267 | 
268 |     ]]>
269 |   </send>
270 | 
271 | </scenario>
272 | 


--------------------------------------------------------------------------------
/dtmf/playdtmf.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="INVITE (optional auth), send some DTMF, Walter Doekes 2016">
  4 | 
  5 |   <!-- Requires SIPp version 3.6+ -->
  6 | 
  7 |   <label id="invite"/>
  8 | 
  9 |   <send retrans="500" start_txn="invite">
 10 |     <![CDATA[
 11 | 
 12 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 13 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 14 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 15 |       To: sip:[tel]@[remote_ip]:[remote_port]
 16 |       Contact: sip:[service]@[local_ip]:[local_port]
 17 |       Call-ID: [call_id]
 18 |       CSeq: [cseq] INVITE
 19 |       Max-Forwards: 70
 20 |       Content-Type: application/sdp
 21 |       Content-Length: [len]
 22 | 
 23 |       v=0
 24 |       o=- 144969 144969 IN IP[local_ip_type] [local_ip]
 25 |       s=-
 26 |       c=IN IP[media_ip_type] [media_ip]
 27 |       t=0 0
 28 |       m=audio [media_port] RTP/AVP 8 0 96
 29 |       a=rtpmap:8 PCMA/8000
 30 |       a=rtpmap:0 PCMU/8000
 31 |       a=rtpmap:96 telephone-event/8000
 32 |       a=fmtp:96 0-16
 33 | 
 34 |     ]]>
 35 |   </send>
 36 | 
 37 |   <recv response="100" optional="true" response_txn="invite"/>
 38 | 
 39 |   <recv response="180" optional="true" response_txn="invite"/>
 40 | 
 41 |   <recv response="181" optional="true" response_txn="invite"/>
 42 | 
 43 |   <recv response="183" optional="true" response_txn="invite"/>
 44 | 
 45 |   <recv response="200" optional="true" rrs="true" response_txn="invite" next="invite-ack"/>
 46 | 
 47 |   <recv response="401" optional="true" next="invite-with-auth" auth="true" response_txn="invite"/>
 48 | 
 49 |   <recv response="407" auth="true" response_txn="invite"/>
 50 | 
 51 |   <label id="invite-with-auth"/>
 52 | 
 53 |   <send ack_txn="invite">
 54 |     <![CDATA[
 55 | 
 56 |       ACK sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 57 |       [last_Via:]
 58 |       [last_From:]
 59 |       [last_To:]
 60 |       Contact: sip:[service]@[local_ip]:[local_port]
 61 |       Call-ID: [call_id]
 62 |       CSeq: [cseq] ACK
 63 |       Max-Forwards: 70
 64 |       Content-Length: 0
 65 | 
 66 |     ]]>
 67 |   </send>
 68 | 
 69 |   <send retrans="500" start_txn="invite">
 70 |     <![CDATA[
 71 | 
 72 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 73 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 74 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 75 |       To: sip:[tel]@[remote_ip]:[remote_port]
 76 |       Contact: sip:[service]@[local_ip]:[local_port]
 77 |       [authentication]
 78 |       Call-ID: [call_id]
 79 |       CSeq: [cseq] INVITE
 80 |       Max-Forwards: 70
 81 |       Content-Type: application/sdp
 82 |       Content-Length: [len]
 83 | 
 84 |       v=0
 85 |       o=- 144969 144969 IN IP[local_ip_type] [local_ip]
 86 |       s=-
 87 |       c=IN IP[media_ip_type] [media_ip]
 88 |       t=0 0
 89 |       m=audio [media_port] RTP/AVP 8 0 96
 90 |       a=rtpmap:8 PCMA/8000
 91 |       a=rtpmap:0 PCMU/8000
 92 |       a=rtpmap:96 telephone-event/8000
 93 |       a=fmtp:96 0-16
 94 | 
 95 |     ]]>
 96 |   </send>
 97 | 
 98 |   <recv response="100" response_txn="invite"/>
 99 | 
100 |   <recv response="180" optional="true" response_txn="invite"/>
101 | 
102 |   <recv response="181" optional="true" response_txn="invite"/>
103 | 
104 |   <recv response="183" optional="true" response_txn="invite"/>
105 | 
106 |   <recv response="200" rrs="true" response_txn="invite"/>
107 | 
108 |   <label id="invite-ack"/>
109 | 
110 |   <send ack_txn="invite">
111 |     <![CDATA[
112 | 
113 |       ACK [next_url] SIP/2.0
114 |       [last_Via:]
115 |       [routes]
116 |       [last_From:]
117 |       [last_To:]
118 |       Contact: sip:[service]@[local_ip]:[local_port]
119 |       Call-ID: [call_id]
120 |       CSeq: [cseq] ACK
121 |       Max-Forwards: 70
122 |       Content-Length: 0
123 | 
124 |     ]]>
125 |   </send>
126 | 
127 |   <nop>
128 |     <action>
129 |       <!-- fire up the rtp stream with a really short beep,
130 |            just in case -->
131 |       <exec play_dtmf="0,50"/>
132 |     </action>
133 |   </nop>
134 |   <pause milliseconds="1500"/>
135 | 
136 |   <nop>
137 |     <action>
138 |       <!--<exec play_dtmf="0123456789*#ABCD,500"/>-->
139 |       <exec play_dtmf="[tones],500"/>
140 |     </action>
141 |   </nop>
142 | 
143 |   <pause milliseconds="18000"/>
144 | 
145 |   <send start_txn="bye">
146 |     <![CDATA[
147 | 
148 |       BYE [next_url] SIP/2.0
149 |       [last_Via:]
150 |       [routes]
151 |       [last_From:]
152 |       [last_To:]
153 |       Contact: sip:[service]@[local_ip]:[local_port]
154 |       Call-ID: [call_id]
155 |       CSeq: [cseq] BYE
156 |       Max-Forwards: 70
157 |       Content-Length: 0
158 | 
159 |     ]]>
160 |   </send>
161 | 
162 |   <recv response="200" response_txn="bye"/>
163 | 
164 | </scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->
165 | 


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/fax/sendfax-wjdtest20150505.pcap:
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https://raw.githubusercontent.com/ossobv/sipp-scenarios/650bde5efa2e887d5184ee5ab571988c43ff648f/fax/sendfax-wjdtest20150505.pcap


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/fax/sendfax.xml:
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  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="INVITE (optional auth), re-INVITE to T38 and send a Fax, Walter Doekes 2012-2013">
  4 | 
  5 |   <!--
  6 | 
  7 |   NOTE: Creating a sendfax.pcap is as easy as:
  8 |   - receive a fax with asterisk
  9 |   - get the incoming side of the spansdp.log
 10 |   - feed it to https://github.com/asterisk/asterisk/blob/master/contrib/scripts/spandspflow2pcap.py
 11 | 
 12 |   -->
 13 | 
 14 |   <label id="invite"/>
 15 | 
 16 |   <send retrans="500" start_txn="invite">
 17 |     <![CDATA[
 18 | 
 19 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 20 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 21 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 22 |       To: sip:[tel]@[remote_ip]:[remote_port]
 23 |       Contact: sip:[service]@[local_ip]:[local_port]
 24 |       Call-ID: [call_id]
 25 |       CSeq: [cseq] INVITE
 26 |       Max-Forwards: 70
 27 |       Content-Type: application/sdp
 28 |       Content-Length: [len]
 29 | 
 30 |       v=0
 31 |       o=- 144969 144969 IN IP[local_ip_type] [local_ip]
 32 |       s=-
 33 |       c=IN IP[media_ip_type] [media_ip]
 34 |       t=0 0
 35 |       m=audio [media_port] RTP/AVP 8 0
 36 |       a=rtpmap:8 PCMA/8000
 37 |       a=rtpmap:0 PCMU/8000
 38 | 
 39 |     ]]>
 40 |   </send>
 41 | 
 42 |   <recv response="100" optional="true" response_txn="invite"/>
 43 | 
 44 |   <recv response="180" optional="true" response_txn="invite"/>
 45 | 
 46 |   <recv response="181" optional="true" response_txn="invite"/>
 47 | 
 48 |   <recv response="183" optional="true" response_txn="invite"/>
 49 | 
 50 |   <recv response="200" optional="true" rrs="true" response_txn="invite" next="invite-ack"/>
 51 | 
 52 |   <recv response="401" optional="true" next="invite-with-auth" auth="true" response_txn="invite"/>
 53 | 
 54 |   <recv response="407" auth="true" response_txn="invite"/>
 55 | 
 56 |   <label id="invite-with-auth"/>
 57 | 
 58 |   <send ack_txn="invite">
 59 |     <![CDATA[
 60 | 
 61 |       ACK sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 62 |       [last_Via:]
 63 |       [last_From:]
 64 |       [last_To:]
 65 |       Contact: sip:[service]@[local_ip]:[local_port]
 66 |       Call-ID: [call_id]
 67 |       CSeq: [cseq] ACK
 68 |       Max-Forwards: 70
 69 |       Content-Length: 0
 70 | 
 71 |     ]]>
 72 |   </send>
 73 | 
 74 |   <send retrans="500" start_txn="invite">
 75 |     <![CDATA[
 76 | 
 77 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 78 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 79 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 80 |       To: sip:[tel]@[remote_ip]:[remote_port]
 81 |       Contact: sip:[service]@[local_ip]:[local_port]
 82 |       [authentication]
 83 |       Call-ID: [call_id]
 84 |       CSeq: [cseq] INVITE
 85 |       Max-Forwards: 70
 86 |       Content-Type: application/sdp
 87 |       Content-Length: [len]
 88 | 
 89 |       v=0
 90 |       o=- 144969 144969 IN IP[local_ip_type] [local_ip]
 91 |       s=-
 92 |       c=IN IP[media_ip_type] [media_ip]
 93 |       t=0 0
 94 |       m=audio [media_port] RTP/AVP 8 0
 95 |       a=rtpmap:8 PCMA/8000
 96 |       a=rtpmap:0 PCMU/8000
 97 | 
 98 |     ]]>
 99 |   </send>
100 | 
101 |   <recv response="100" response_txn="invite"/>
102 | 
103 |   <recv response="180" optional="true" response_txn="invite"/>
104 | 
105 |   <recv response="181" optional="true" response_txn="invite"/>
106 | 
107 |   <recv response="183" optional="true" response_txn="invite"/>
108 | 
109 |   <recv response="200" rrs="true" response_txn="invite"/>
110 | 
111 |   <label id="invite-ack"/>
112 | 
113 |   <send ack_txn="invite">
114 |     <![CDATA[
115 | 
116 |       ACK [next_url] SIP/2.0
117 |       [last_Via:]
118 |       [routes]
119 |       [last_From:]
120 |       [last_To:]
121 |       Contact: sip:[service]@[local_ip]:[local_port]
122 |       Call-ID: [call_id]
123 |       CSeq: [cseq] ACK
124 |       Max-Forwards: 70
125 |       Content-Length: 0
126 | 
127 |     ]]>
128 |   </send>
129 | 
130 |   <!-- Send a bit of noise to start the RTP. You may need to use
131 |        the -i MY_IP command line option. -->
132 |   <pause milliseconds="500"/>
133 | 
134 |   <nop>
135 |     <action>
136 |       <exec play_pcap_audio="g711a.pcap"/>
137 |     </action>
138 |   </nop>
139 | 
140 |   <pause milliseconds="500"/>
141 | 
142 | 
143 |   <!-- *****************************************************************
144 | 
145 |   Initiate re-INVITE to T38.
146 | 
147 |   ****************************************************************** -->
148 | 
149 |   <send start_txn="t38invite">
150 |     <![CDATA[
151 | 
152 |       INVITE [next_url] SIP/2.0
153 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
154 |       [routes]
155 |       [last_From:]
156 |       [last_To:]
157 |       Contact: sip:[service]@[local_ip]:[local_port]
158 |       Call-ID: [call_id]
159 |       CSeq: [cseq] INVITE
160 |       Max-Forwards: 70
161 |       Content-Type: application/sdp
162 |       Content-Length: [len]
163 | 
164 |       v=0
165 |       o=- 145280 145280 IN IP[local_ip_type] [local_ip]
166 |       s=-
167 |       c=IN IP[media_ip_type] [media_ip]
168 |       t=0 0
169 |       m=image [media_port] udptl t38
170 |       a=T38FaxVersion:0
171 |       a=T38MaxBitRate:14400
172 |       a=T38FaxRateManagement:transferredTCF
173 |       a=T38FaxMaxBuffer:200
174 |       a=T38FaxMaxDatagram:200
175 |       a=T38FaxUdpEC:t38UDPRedundancy
176 | 
177 |     ]]>
178 |   </send>
179 | 
180 |   <recv response="100" optional="true" response_txn="t38invite"/>
181 | 
182 |   <recv response="488" optional="true" response_txn="t38invite" next="abort"/>
183 | 
184 |   <recv response="200" response_txn="t38invite"/>
185 | 
186 |   <send ack_txn="t38invite">
187 |     <![CDATA[
188 | 
189 |       ACK [next_url] SIP/2.0
190 |       [last_Via:]
191 |       [routes]
192 |       [last_From:]
193 |       [last_To:]
194 |       Contact: sip:[service]@[local_ip]:[local_port]
195 |       Call-ID: [call_id]
196 |       CSeq: [cseq] ACK
197 |       Max-Forwards: 70
198 |       Content-Length: 0
199 | 
200 |     ]]>
201 |   </send>
202 | 
203 |   <!-- Enable the nop/action below to replay the FAX image. You may need to use
204 |        the -i MY_IP command line option. -->
205 |   <pause milliseconds="500"/>
206 | 
207 |   <nop>
208 |     <action>
209 |       <exec play_pcap_image="sendfax-wjdtest20150505.pcap"/>
210 |     </action>
211 |   </nop>
212 | 
213 | 
214 |   <!-- *****************************************************************
215 | 
216 |   Wait for re-INVITE back to audio.
217 | 
218 |   ****************************************************************** -->
219 | 
220 |   <recv request="INVITE"/>
221 | 
222 |   <send retrans="500">
223 |     <![CDATA[
224 | 
225 |       SIP/2.0 200 OK
226 |       [last_Via:]
227 |       [last_Record-Route:]
228 |       [last_From:]
229 |       [last_To:]
230 |       Contact: sip:[service]@[local_ip]:[local_port]
231 |       [last_Call-ID:]
232 |       [last_CSeq:]
233 |       Max-Forwards: 70
234 |       Content-Type: application/sdp
235 |       Content-Length: [len]
236 | 
237 |       v=0
238 |       o=- 146312 146312 IN IP[local_ip_type] [local_ip]
239 |       s=-
240 |       c=IN IP[media_ip_type] [media_ip]
241 |       t=0 0
242 |       m=audio [media_port] RTP/AVP 8 0
243 |       a=rtpmap:8 PCMA/8000
244 |       a=rtpmap:0 PCMU/8000
245 | 
246 |     ]]>
247 |   </send>
248 | 
249 |   <recv request="ACK"/>
250 | 
251 |   <recv request="BYE"/>
252 | 
253 |   <send next="done">
254 |     <![CDATA[
255 | 
256 |       SIP/2.0 200 OK
257 |       [last_Via:]
258 |       [last_Record-Route:]
259 |       [last_From:]
260 |       [last_To:]
261 |       Contact: sip:[service]@[local_ip]:[local_port]
262 |       [last_Call-ID:]
263 |       [last_CSeq:]
264 |       Content-Length: 0
265 | 
266 |     ]]>
267 |   </send>
268 | 
269 | 
270 |   <!-- *****************************************************************
271 | 
272 |   Abort the call ourselves
273 | 
274 |   ****************************************************************** -->
275 | 
276 |   <label id="abort"/>
277 | 
278 |   <send ack_txn="t38invite">
279 |     <![CDATA[
280 | 
281 |       ACK [next_url] SIP/2.0
282 |       [last_Via:]
283 |       [routes]
284 |       [last_From:]
285 |       [last_To:]
286 |       Contact: sip:[service]@[local_ip]:[local_port]
287 |       Call-ID: [call_id]
288 |       CSeq: [cseq] ACK
289 |       Max-Forwards: 70
290 |       Content-Length: 0
291 | 
292 |     ]]>
293 |   </send>
294 | 
295 |   <send start_txn="bye">
296 |     <![CDATA[
297 | 
298 |       BYE [next_url] SIP/2.0
299 |       [last_Via:]
300 |       [routes]
301 |       [last_From:]
302 |       [last_To:]
303 |       Contact: sip:[service]@[local_ip]:[local_port]
304 |       Call-ID: [call_id]
305 |       CSeq: [cseq] BYE
306 |       Max-Forwards: 70
307 |       Content-Length: 0
308 | 
309 |     ]]>
310 |   </send>
311 | 
312 |   <recv response="200" response_txn="bye" next="done"/>
313 | 
314 | 
315 |   <!-- *****************************************************************
316 | 
317 |   Finalize
318 | 
319 |   ****************************************************************** -->
320 | 
321 |   <label id="done"/>
322 | 
323 |   <!-- Keep call open to be able to retransmit stuff -->
324 |   <timewait milliseconds="2000"/>
325 | 
326 | </scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->
327 | 


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/out-in/beep_1sec_50x160b.alaw:
--------------------------------------------------------------------------------
https://raw.githubusercontent.com/ossobv/sipp-scenarios/650bde5efa2e887d5184ee5ab571988c43ff648f/out-in/beep_1sec_50x160b.alaw


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/out-in/i3pcc-oinv-o3pcc-i603.xml:
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  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="i3pcc-oinv-o3pcc-i603.xml -- wjd 2015">
  4 | 
  5 |   <!--
  6 | 
  7 |   * Wait for 3pcc poke.
  8 |   * Invite, wait for ringing.
  9 |   * Tell 3pcc about our dialog.
 10 |   * Cancel this call.
 11 | 
 12 |   -->
 13 | 
 14 |   <!-- Wait for poke before doing anything.
 15 |        Note that we auto-use the Call-ID we get from this 3pcc connection.
 16 |        We don't want that, because this is a different call: hence we
 17 |        prepend "dedup///" everywhere. (MUST be with 3 slashes, or sipp
 18 |        will not recognise that this call_id belongs to this sipp
 19 |        instance. -->
 20 |   <recvCmd/>
 21 | 
 22 |   <send retrans="500" start_txn="invite">
 23 |     <![CDATA[
 24 | 
 25 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 26 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 27 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 28 |       To: sip:[tel]@[remote_ip]:[remote_port]
 29 |       Call-ID: dedup///[call_id]
 30 |       CSeq: [cseq] INVITE
 31 |       Contact: sip:[service]@[local_ip]:[local_port]
 32 |       Content-Type: application/sdp
 33 |       Content-Length: [len]
 34 | 
 35 |       v=0
 36 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 37 |       s=-
 38 |       c=IN IP[media_ip_type] [media_ip]
 39 |       t=0 0
 40 |       m=audio [media_port] RTP/AVP 8 0
 41 |       a=rtpmap:8 PCMA/8000
 42 |       a=rtpmap:0 PCMU/8000
 43 | 
 44 |     ]]>
 45 |   </send>
 46 | 
 47 |   <recv response="100" optional="true" response_txn="invite"/>
 48 | 
 49 |   <recv response="401" optional="true" auth="true" next="invite-with-auth" response_txn="invite">
 50 |     <action>
 51 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 52 |     </action>
 53 |   </recv>
 54 | 
 55 |   <recv response="407" auth="true" response_txn="invite">
 56 |     <action>
 57 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 58 |     </action>
 59 |   </recv>
 60 | 
 61 |   <label id="invite-with-auth"/>
 62 | 
 63 |   <!-- ACK is within transaction because of non-2xx. Reuse Via branch.      -->
 64 |   <!-- We don't use last_Via: because it would include rport stuff from     -->
 65 |   <!-- remote.                                                              -->
 66 |   <send ack_txn="invite">
 67 |     <![CDATA[
 68 | 
 69 |       ACK sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 70 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$1]
 71 |       [last_From:]
 72 |       [last_To:]
 73 |       [last_Call-ID:]
 74 |       CSeq: [cseq] ACK
 75 |       Contact: sip:[service]@[local_ip]:[local_port]
 76 |       Content-Length: 0
 77 | 
 78 |     ]]>
 79 |   </send>
 80 | 
 81 |   <send retrans="500" start_txn="invite">
 82 |     <![CDATA[
 83 | 
 84 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 85 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 86 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 87 |       To: sip:[tel]@[remote_ip]:[remote_port]
 88 |       [authentication]
 89 |       Call-ID: dedup///[call_id]
 90 |       CSeq: [cseq] INVITE
 91 |       Contact: sip:[service]@[local_ip]:[local_port]
 92 |       Content-Type: application/sdp
 93 |       Content-Length: [len]
 94 | 
 95 |       v=0
 96 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 97 |       s=-
 98 |       c=IN IP[media_ip_type] [media_ip]
 99 |       t=0 0
100 |       m=audio [media_port] RTP/AVP 8 0
101 |       a=rtpmap:8 PCMA/8000
102 |       a=rtpmap:0 PCMU/8000
103 | 
104 |     ]]>
105 |   </send>
106 | 
107 |   <recv response="100" response_txn="invite"/>
108 | 
109 |   <recv response="180" response_txn="invite"/>
110 | 
111 |   <!-- Send dialog credentials over 3pcc.
112 |        There is no next_url to use yet. Use the last_Contact instead.
113 |        (Call-ID must be in there.) -->
114 |   <sendCmd>
115 |     <![CDATA[
116 | 
117 |       [last_From:]
118 |       [last_To:]
119 |       Call-ID: [call_id]
120 |       [last_Contact:]
121 | 
122 |     ]]>
123 |   </sendCmd>
124 | 
125 |   <!-- This call has been replaced and should be cancelled.
126 |        Oddly enough, Asterisk does this by sending a 603. -->
127 |   <recv response="603" response_txn="invite">
128 |     <action>
129 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
130 |     </action>
131 |   </recv>
132 | 
133 |   <!-- ACK is within transaction because of non-2xx. Reuse Via branch.      -->
134 |   <!-- We don't use last_Via: because it would include rport stuff from     -->
135 |   <!-- remote.                                                              -->
136 |   <send ack_txn="invite">
137 |     <![CDATA[
138 | 
139 |       ACK sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
140 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$1]
141 |       [last_From:]
142 |       [last_To:]
143 |       [last_Call-ID:]
144 |       CSeq: [cseq] ACK
145 |       Contact: sip:[service]@[local_ip]:[local_port]
146 |       Content-Length: 0
147 | 
148 |     ]]>
149 |   </send>
150 | 
151 |   <timewait milliseconds="2000"/><!-- retransmit ACKs for a while -->
152 | 
153 | </scenario>
154 | 


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/out-in/i3pcc-oinv-o3pcc-ibye.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="i3pcc-oinv-o3pcc-ibye.xml -- wjd 2015">
  4 | 
  5 |   <!--
  6 | 
  7 |   * Wait for 3pcc poke.
  8 |   * Invite.
  9 |   * Tell 3pcc about our dialog.
 10 |   * Wait for the call hangup.
 11 | 
 12 |   -->
 13 | 
 14 |   <!-- Wait for poke before doing anything.
 15 |        Note that we auto-use the Call-ID we get from this 3pcc connection.
 16 |        We don't want that, because this is a different call: hence we
 17 |        prepend "dedup///" everywhere. (MUST be with 3 slashes, or sipp
 18 |        will not recognise that this call_id belongs to this sipp
 19 |        instance. -->
 20 |   <recvCmd/>
 21 | 
 22 |   <send retrans="500" start_txn="invite">
 23 |     <![CDATA[
 24 | 
 25 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 26 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 27 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 28 |       To: sip:[tel]@[remote_ip]:[remote_port]
 29 |       Call-ID: dedup///[call_id]
 30 |       CSeq: [cseq] INVITE
 31 |       Contact: sip:[service]@[local_ip]:[local_port]
 32 |       Content-Type: application/sdp
 33 |       Content-Length: [len]
 34 | 
 35 |       v=0
 36 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 37 |       s=-
 38 |       c=IN IP[media_ip_type] [media_ip]
 39 |       t=0 0
 40 |       m=audio [media_port] RTP/AVP 8 0
 41 |       a=rtpmap:8 PCMA/8000
 42 |       a=rtpmap:0 PCMU/8000
 43 | 
 44 |     ]]>
 45 |   </send>
 46 | 
 47 |   <recv response="100" optional="true" response_txn="invite"/>
 48 | 
 49 |   <recv response="401" optional="true" auth="true" next="invite-with-auth" response_txn="invite">
 50 |     <action>
 51 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 52 |     </action>
 53 |   </recv>
 54 | 
 55 |   <recv response="407" auth="true" response_txn="invite">
 56 |     <action>
 57 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 58 |     </action>
 59 |   </recv>
 60 | 
 61 |   <label id="invite-with-auth"/>
 62 | 
 63 |   <!-- ACK is within transaction because of non-2xx. Reuse Via branch.      -->
 64 |   <!-- We don't use last_Via: because it would include rport stuff from     -->
 65 |   <!-- remote.                                                              -->
 66 |   <send ack_txn="invite">
 67 |     <![CDATA[
 68 | 
 69 |       ACK sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 70 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$1]
 71 |       [last_From:]
 72 |       [last_To:]
 73 |       [last_Call-ID:]
 74 |       CSeq: [cseq] ACK
 75 |       Contact: sip:[service]@[local_ip]:[local_port]
 76 |       Content-Length: 0
 77 | 
 78 |     ]]>
 79 |   </send>
 80 | 
 81 |   <send retrans="500" start_txn="invite">
 82 |     <![CDATA[
 83 | 
 84 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 85 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 86 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 87 |       To: sip:[tel]@[remote_ip]:[remote_port]
 88 |       [authentication]
 89 |       Call-ID: dedup///[call_id]
 90 |       CSeq: [cseq] INVITE
 91 |       Contact: sip:[service]@[local_ip]:[local_port]
 92 |       Content-Type: application/sdp
 93 |       Content-Length: [len]
 94 | 
 95 |       v=0
 96 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 97 |       s=-
 98 |       c=IN IP[media_ip_type] [media_ip]
 99 |       t=0 0
100 |       m=audio [media_port] RTP/AVP 8 0
101 |       a=rtpmap:8 PCMA/8000
102 |       a=rtpmap:0 PCMU/8000
103 | 
104 |     ]]>
105 |   </send>
106 | 
107 |   <recv response="100" response_txn="invite"/>
108 | 
109 |   <recv response="180" optional="true" response_txn="invite"/>
110 | 
111 |   <recv response="183" optional="true" response_txn="invite"/>
112 | 
113 |   <recv response="200" rrs="true" response_txn="invite"/>
114 | 
115 |   <send ack_txn="invite">
116 |     <![CDATA[
117 | 
118 |       ACK [next_url] SIP/2.0
119 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
120 |       [routes]
121 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
122 |       To: sip:[tel]@[remote_ip]:[remote_port][peer_tag_param]
123 |       Call-ID: dedup///[call_id]
124 |       CSeq: [cseq] ACK
125 |       Contact: sip:[service]@[local_ip]:[local_port]
126 |       Content-Length: 0
127 | 
128 |     ]]>
129 |   </send>
130 | 
131 |   <nop>
132 |     <action>
133 |       <exec rtp_stream="beep_1sec_50x160b.alaw,1,8"/>
134 |     </action>
135 |   </nop>
136 |   <pause milliseconds="1000"/>
137 | 
138 |   <!-- Send dialog credentials over 3pcc.
139 |        (Call-ID must be in there.) -->
140 |   <sendCmd>
141 |     <![CDATA[
142 | 
143 |       [last_From:]
144 |       [last_To:]
145 |       Call-ID: [call_id]
146 |       Contact: [next_url]
147 | 
148 |     ]]>
149 |   </sendCmd>
150 | 
151 |   <!-- This call has been replaced and will be cancelled. -->
152 |   <recv request="BYE"/>
153 | 
154 |   <send>
155 |     <![CDATA[
156 | 
157 |       SIP/2.0 200 OK
158 |       [last_Via:]
159 |       [last_Record-Route:]
160 |       [last_From:]
161 |       [last_To:]
162 |       [last_Call-ID:]
163 |       [last_CSeq:]
164 |       Content-Length: 0
165 | 
166 |     ]]>
167 |   </send>
168 | 
169 |   <timewait milliseconds="2000"/><!-- retransmit 200's for a while -->
170 | 
171 | </scenario>
172 | 


--------------------------------------------------------------------------------
/out-in/iinv-o180-icancel.xml:
--------------------------------------------------------------------------------
 1 | <?xml version="1.0" encoding="UTF-8" ?>
 2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
 3 | <scenario name="iinv-o180-icancel.xml -- wjd 2015">
 4 | 
 5 |   <recv request="INVITE" rrs="true">
 6 |     <action>
 7 |       <ereg occurrence="1" regexp=": .*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 8 |       <ereg occurrence="2" regexp=": .*" search_in="hdr" header="Via" check_it="false" assign_to="2"/>
 9 |       <ereg regexp=": .*" search_in="hdr" header="Record-Route" check_it="false" assign_to="3"/>
10 |       <ereg regexp=": .*" search_in="hdr" header="CSeq" check_it="false" assign_to="4"/>
11 |     </action>
12 |   </recv>
13 | 
14 |   <send>
15 |     <![CDATA[
16 | 
17 |       SIP/2.0 180 Ringing
18 |       [last_Via:]
19 |       [last_Record-Route:]
20 |       [last_From:]
21 |       [last_To:];tag=[pid]SIPpTag01[call_number]
22 |       [last_Call-ID:]
23 |       [last_CSeq:]
24 |       Contact: sip:[service]@[local_ip]:[local_port]
25 |       Content-Length: [len]
26 | 
27 |     ]]>
28 |   </send>
29 | 
30 |   <recv request="CANCEL"/>
31 | 
32 |   <send>
33 |     <![CDATA[
34 | 
35 |       SIP/2.0 200 OK
36 |       [last_Via:]
37 |       [last_Record-Route:]
38 |       [last_From:]
39 |       [last_To:]
40 |       [last_Call-ID:]
41 |       [last_CSeq:]
42 |       Content-Length: 0
43 | 
44 |     ]]>
45 |   </send>
46 | 
47 |   <send>
48 |     <![CDATA[
49 | 
50 |       SIP/2.0 487 Request Terminated
51 |       Via[$1]
52 |       Via[$2]
53 |       Record-Route[$3]
54 |       [last_From:]
55 |       [last_To:];tag=[pid]SIPpTag01[call_number]
56 |       [last_Call-ID:]
57 |       CSeq[$4]
58 |       Content-Length: 0
59 | 
60 |     ]]>
61 |   </send>
62 | 
63 |   <recv request="ACK"/>
64 | 
65 | </scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->
66 | 


--------------------------------------------------------------------------------
/out-in/iinv-o180-wait-o200-ibye.xml:
--------------------------------------------------------------------------------
 1 | <?xml version="1.0" encoding="UTF-8" ?>
 2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
 3 | <scenario name="iinv-o180-wait-o200-ibye -- wjd 2015">
 4 | 
 5 |   <!--
 6 | 
 7 |   * Regular "wait for invite and send bye".
 8 |   * Wait for 180wait key.
 9 | 
10 |   -->
11 | 
12 |   <recv request="INVITE" rrs="true"/>
13 | 
14 |   <send>
15 |     <![CDATA[
16 | 
17 |       SIP/2.0 180 Ringing
18 |       [last_Via:]
19 |       [last_Record-Route:]
20 |       [last_From:]
21 |       [last_To:];tag=[pid]SIPpTag01[call_number]
22 |       [last_Call-ID:]
23 |       [last_CSeq:]
24 |       Contact: sip:[service]@[local_ip]:[local_port]
25 |       Content-Length: 0
26 | 
27 |     ]]>
28 |   </send>
29 | 
30 |   <!-- Use 180wait key to optionally wait a while. -->
31 |   <!-- Ouch! This is fugly code right there, just to wait a while... -->
32 |   <nop hide="true"><action>
33 |     <assignstr assign_to="tmp" value="[180wait]"/>
34 |     <todouble assign_to="tmpi" variable="tmp"/></action></nop>
35 |   <label id="restart180"/>
36 |   <pause milliseconds="500"/>
37 |   <nop hide="true"><action>
38 |     <subtract assign_to="tmpi" value="1"/>
39 |     <test assign_to="tmp" variable="tmpi" compare="greater_than" value="0"/></action></nop>
40 |   <nop next="restart180" test="tmp"/>
41 | 
42 |   <send retrans="500">
43 |     <![CDATA[
44 | 
45 |       SIP/2.0 200 OK
46 |       [last_Via:]
47 |       [last_Record-Route:]
48 |       [last_From:]
49 |       [last_To:];tag=[pid]SIPpTag01[call_number]
50 |       [last_Call-ID:]
51 |       [last_CSeq:]
52 |       Contact: sip:[service]@[local_ip]:[local_port]
53 |       Content-Type: application/sdp
54 |       Content-Length: [len]
55 | 
56 |       v=0
57 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
58 |       s=-
59 |       c=IN IP[media_ip_type] [media_ip]
60 |       t=0 0
61 |       m=audio [media_port] RTP/AVP 8 0
62 |       a=rtpmap:8 PCMA/8000
63 |       a=rtpmap:0 PCMU/8000
64 | 
65 |     ]]>
66 |   </send>
67 | 
68 |   <recv request="ACK"/>
69 | 
70 |   <nop>
71 |     <action>
72 |       <exec rtp_stream="beep_1sec_50x160b.alaw,1,8"/>
73 |     </action>
74 |   </nop>
75 | 
76 |   <!-- We're done. Wait for the peer to hang up. -->
77 |   <recv request="BYE"/>
78 | 
79 |   <send>
80 |     <![CDATA[
81 | 
82 |       SIP/2.0 200 OK
83 |       [last_Via:]
84 |       [last_Record-Route:]
85 |       [last_From:]
86 |       [last_To:]
87 |       [last_Call-ID:]
88 |       [last_CSeq:]
89 |       Content-Length: 0
90 | 
91 |     ]]>
92 |   </send>
93 | 
94 |   <timewait milliseconds="2000"/><!-- retransmit 200's for a while -->
95 | 
96 | </scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->
97 | 


--------------------------------------------------------------------------------
/out-in/iinv-o180-wait-o200-iinv-o200-obye.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="iinv-o180-wait-o200-iinv-o200-obye.xml -- wjd 2015">
  4 | 
  5 |   <!--
  6 | 
  7 |   * Regular "wait for invite and send bye".
  8 |   * Wait for 180wait key.
  9 |   * Expect a reinvite after we 200.
 10 | 
 11 |   -->
 12 | 
 13 |   <recv request="INVITE" rrs="true"/>
 14 | 
 15 |   <send>
 16 |     <![CDATA[
 17 | 
 18 |       SIP/2.0 180 Ringing
 19 |       [last_Via:]
 20 |       [last_Record-Route:]
 21 |       [last_From:]
 22 |       [last_To:];tag=[pid]SIPpTag01[call_number]
 23 |       [last_Call-ID:]
 24 |       [last_CSeq:]
 25 |       Contact: sip:[service]@[local_ip]:[local_port]
 26 |       Content-Length: 0
 27 | 
 28 |     ]]>
 29 |   </send>
 30 | 
 31 |   <!-- At this point, we wait a while. On the other side, transfers are
 32 |        happening. -->
 33 | 
 34 |   <!-- Use 180wait key to optionally wait a while. -->
 35 |   <!-- Ouch! This is fugly code right there, just to wait a while... -->
 36 |   <nop hide="true"><action>
 37 |     <assignstr assign_to="tmp" value="[180wait]"/>
 38 |     <todouble assign_to="tmpi" variable="tmp"/></action></nop>
 39 |   <label id="restart180"/>
 40 |   <pause milliseconds="500"/>
 41 |   <nop hide="true"><action>
 42 |     <subtract assign_to="tmpi" value="1"/>
 43 |     <test assign_to="tmp" variable="tmpi" compare="greater_than" value="0"/></action></nop>
 44 |   <nop next="restart180" test="tmp"/>
 45 | 
 46 |   <send retrans="500">
 47 |     <![CDATA[
 48 | 
 49 |       SIP/2.0 200 OK
 50 |       [last_Via:]
 51 |       [last_Record-Route:]
 52 |       [last_From:]
 53 |       [last_To:];tag=[pid]SIPpTag01[call_number]
 54 |       [last_Call-ID:]
 55 |       [last_CSeq:]
 56 |       Contact: sip:[service]@[local_ip]:[local_port]
 57 |       Content-Type: application/sdp
 58 |       Content-Length: [len]
 59 | 
 60 |       v=0
 61 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 62 |       s=-
 63 |       c=IN IP[media_ip_type] [media_ip]
 64 |       t=0 0
 65 |       m=audio [media_port] RTP/AVP 8 0
 66 |       a=rtpmap:8 PCMA/8000
 67 |       a=rtpmap:0 PCMU/8000
 68 | 
 69 |     ]]>
 70 |   </send>
 71 | 
 72 |   <!-- Send ACK and wait for reINVITE because of transfer on other end.
 73 |        The transfer causes an immediate update by Asterisk, which in
 74 |        rare cases can get reordered. We allow it, but wait for the
 75 |        retransmit to answer it. -->
 76 |   <recv request="INVITE" optional="true"/>
 77 |   <recv request="ACK"/>
 78 |   <recv request="INVITE" rrs="true"/>
 79 | 
 80 |   <!-- Reply with unchanged session -->
 81 |   <send retrans="500">
 82 |     <![CDATA[
 83 | 
 84 |       SIP/2.0 200 OK
 85 |       [last_Via:]
 86 |       [last_Record-Route:]
 87 |       [last_From:]
 88 |       [last_To:]
 89 |       [last_Call-ID:]
 90 |       [last_CSeq:]
 91 |       Contact: sip:[service]@[local_ip]:[local_port]
 92 |       Content-Type: application/sdp
 93 |       Content-Length: [len]
 94 | 
 95 |       v=0
 96 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 97 |       s=-
 98 |       c=IN IP[media_ip_type] [media_ip]
 99 |       t=0 0
100 |       m=audio [media_port] RTP/AVP 8 0
101 |       a=rtpmap:8 PCMA/8000
102 |       a=rtpmap:0 PCMU/8000
103 | 
104 |     ]]>
105 |   </send>
106 | 
107 |   <recv request="ACK">
108 |     <action>
109 |       <ereg regexp=": *(.*)" search_in="hdr" header="To" assign_to="_,us"/><!-- with tag -->
110 |       <ereg regexp=": *(.*)" search_in="hdr" header="From" assign_to="_,them"/>
111 |     </action>
112 |   </recv>
113 |   <Reference variables="_"/>
114 | 
115 |   <!-- Sleep a short while, keeping the new session open. -->
116 |   <nop>
117 |     <action>
118 |       <exec rtp_stream="beep_1sec_50x160b.alaw,1,8"/>
119 |     </action>
120 |   </nop>
121 |   <pause milliseconds="1000"/>
122 | 
123 |   <!-- We're done. Hang up. -->
124 |   <send start_txn="bye">
125 |     <![CDATA[
126 | 
127 |       BYE [next_url] SIP/2.0
128 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
129 |       [routes]
130 |       From: [$us]
131 |       To: [$them]
132 |       [last_Call-ID:]
133 |       CSeq: [cseq] BYE
134 |       Contact: sip:[service]@[local_ip]:[local_port]
135 |       Content-Length: 0
136 | 
137 |     ]]>
138 |   </send>
139 | 
140 |   <recv response="200" response_txn="bye"/>
141 | 
142 | </scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->
143 | 


--------------------------------------------------------------------------------
/out-in/iinv-o180-wait-o200-wait-obye.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="iinv-o180-wait-o200-wait-obye.xml -- wjd 2015">
  4 | 
  5 |   <!--
  6 | 
  7 |   * Regular "wait for invite and send bye".
  8 |   * Wait for 180wait key and 200wait key.
  9 | 
 10 |   -->
 11 | 
 12 |   <recv request="INVITE" rrs="true"/>
 13 | 
 14 |   <send>
 15 |     <![CDATA[
 16 | 
 17 |       SIP/2.0 180 Ringing
 18 |       [last_Via:]
 19 |       [last_Record-Route:]
 20 |       [last_From:]
 21 |       [last_To:];tag=[pid]SIPpTag01[call_number]
 22 |       [last_Call-ID:]
 23 |       [last_CSeq:]
 24 |       Contact: sip:[service]@[local_ip]:[local_port]
 25 |       Content-Length: 0
 26 | 
 27 |     ]]>
 28 |   </send>
 29 | 
 30 |   <!-- Use 180wait key to optionally wait a while. -->
 31 |   <!-- Ouch! This is fugly code right there, just to wait a while... -->
 32 |   <nop hide="true"><action>
 33 |     <assignstr assign_to="tmp" value="[180wait]"/>
 34 |     <todouble assign_to="tmpi" variable="tmp"/></action></nop>
 35 |   <label id="restart180"/>
 36 |   <pause milliseconds="500"/>
 37 |   <nop hide="true"><action>
 38 |     <subtract assign_to="tmpi" value="1"/>
 39 |     <test assign_to="tmp" variable="tmpi" compare="greater_than" value="0"/></action></nop>
 40 |   <nop next="restart180" test="tmp"/>
 41 | 
 42 |   <send retrans="500">
 43 |     <![CDATA[
 44 | 
 45 |       SIP/2.0 200 OK
 46 |       [last_Via:]
 47 |       [last_Record-Route:]
 48 |       [last_From:]
 49 |       [last_To:];tag=[pid]SIPpTag01[call_number]
 50 |       [last_Call-ID:]
 51 |       [last_CSeq:]
 52 |       Contact: sip:[service]@[local_ip]:[local_port]
 53 |       Content-Type: application/sdp
 54 |       Content-Length: [len]
 55 | 
 56 |       v=0
 57 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 58 |       s=-
 59 |       c=IN IP[media_ip_type] [media_ip]
 60 |       t=0 0
 61 |       m=audio [media_port] RTP/AVP 8 0
 62 |       a=rtpmap:8 PCMA/8000
 63 |       a=rtpmap:0 PCMU/8000
 64 | 
 65 |     ]]>
 66 |   </send>
 67 | 
 68 |   <recv request="ACK">
 69 |     <action>
 70 |       <ereg regexp=": *(.*)" search_in="hdr" header="To" assign_to="_,us"/><!-- with tag -->
 71 |       <ereg regexp=": *(.*)" search_in="hdr" header="From" assign_to="_,them"/>
 72 |     </action>
 73 |   </recv>
 74 |   <Reference variables="_"/>
 75 | 
 76 |   <!-- At this point, we wait a while. On the other side, transfers are
 77 |        happening. -->
 78 |   <nop>
 79 |     <action>
 80 |       <exec rtp_stream="beep_1sec_50x160b.alaw,4,8"/>
 81 |     </action>
 82 |   </nop>
 83 | 
 84 |   <!-- Use 200wait key to optionally wait a while. -->
 85 |   <!-- Ouch! This is fugly code right there, just to wait a while... -->
 86 |   <nop hide="true"><action>
 87 |     <assignstr assign_to="tmp" value="[200wait]"/>
 88 |     <todouble assign_to="tmpi" variable="tmp"/></action></nop>
 89 |   <label id="restart200"/>
 90 |   <pause milliseconds="500"/>
 91 |   <nop hide="true"><action>
 92 |     <subtract assign_to="tmpi" value="1"/>
 93 |     <test assign_to="tmp" variable="tmpi" compare="greater_than" value="0"/></action></nop>
 94 |   <nop next="restart200" test="tmp"/>
 95 | 
 96 |   <send start_txn="bye">
 97 |     <![CDATA[
 98 | 
 99 |       BYE [next_url] SIP/2.0
100 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
101 |       [routes]
102 |       From: [$us]
103 |       To: [$them]
104 |       [last_Call-ID:]
105 |       CSeq: [cseq] BYE
106 |       Contact: sip:[service]@[local_ip]:[local_port]
107 |       Content-Length: 0
108 | 
109 |     ]]>
110 |   </send>
111 | 
112 |   <recv response="200" response_txn="bye"/>
113 | 
114 | </scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->
115 | 


--------------------------------------------------------------------------------
/out-in/iinv-o200-iinv-o200-ibye.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="iinv-o200-iinv-o200-ibye.xml -- wjd 2015">
  4 | 
  5 |   <!--
  6 | 
  7 |   * Regular "wait for invite and send bye".
  8 |   * Except we expect a reinvite too.
  9 | 
 10 |   -->
 11 | 
 12 |   <recv request="INVITE" rrs="true"/>
 13 | 
 14 |   <send>
 15 |     <![CDATA[
 16 | 
 17 |       SIP/2.0 180 Ringing
 18 |       [last_Via:]
 19 |       [last_Record-Route:]
 20 |       [last_From:]
 21 |       [last_To:];tag=[pid]SIPpTag01[call_number]
 22 |       [last_Call-ID:]
 23 |       [last_CSeq:]
 24 |       Contact: sip:[service]@[local_ip]:[local_port]
 25 |       Content-Length: 0
 26 | 
 27 |     ]]>
 28 |   </send>
 29 | 
 30 |   <send retrans="500">
 31 |     <![CDATA[
 32 | 
 33 |       SIP/2.0 200 OK
 34 |       [last_Via:]
 35 |       [last_Record-Route:]
 36 |       [last_From:]
 37 |       [last_To:];tag=[pid]SIPpTag01[call_number]
 38 |       [last_Call-ID:]
 39 |       [last_CSeq:]
 40 |       Contact: sip:[service]@[local_ip]:[local_port]
 41 |       Content-Type: application/sdp
 42 |       Content-Length: [len]
 43 | 
 44 |       v=0
 45 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 46 |       s=-
 47 |       c=IN IP[media_ip_type] [media_ip]
 48 |       t=0 0
 49 |       m=audio [media_port] RTP/AVP 8 0
 50 |       a=rtpmap:8 PCMA/8000
 51 |       a=rtpmap:0 PCMU/8000
 52 | 
 53 |     ]]>
 54 |   </send>
 55 | 
 56 |   <recv request="ACK"/>
 57 | 
 58 |   <nop>
 59 |     <action>
 60 |       <exec rtp_stream="beep_1sec_50x160b.alaw,1,8"/>
 61 |     </action>
 62 |   </nop>
 63 | 
 64 |   <!-- Wait for reINVITE because of transfer on other end. -->
 65 |   <recv request="INVITE" rrs="true"/>
 66 | 
 67 |   <!-- Reply with unchanged session -->
 68 |   <send retrans="500">
 69 |     <![CDATA[
 70 | 
 71 |       SIP/2.0 200 OK
 72 |       [last_Via:]
 73 |       [last_Record-Route:]
 74 |       [last_From:]
 75 |       [last_To:]
 76 |       [last_Call-ID:]
 77 |       [last_CSeq:]
 78 |       Contact: sip:[service]@[local_ip]:[local_port]
 79 |       Content-Type: application/sdp
 80 |       Content-Length: [len]
 81 | 
 82 |       v=0
 83 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 84 |       s=-
 85 |       c=IN IP[media_ip_type] [media_ip]
 86 |       t=0 0
 87 |       m=audio [media_port] RTP/AVP 8 0
 88 |       a=rtpmap:8 PCMA/8000
 89 |       a=rtpmap:0 PCMU/8000
 90 | 
 91 |     ]]>
 92 |   </send>
 93 | 
 94 |   <recv request="ACK"/>
 95 | 
 96 |   <!-- Resume sending a bit of audio. -->
 97 |   <nop>
 98 |     <action>
 99 |       <exec rtp_stream="beep_1sec_50x160b.alaw,1,8"/>
100 |     </action>
101 |   </nop>
102 | 
103 |   <!-- We're done. Wait for the peer to hang up. -->
104 |   <recv request="BYE"/>
105 | 
106 |   <send>
107 |     <![CDATA[
108 | 
109 |       SIP/2.0 200 OK
110 |       [last_Via:]
111 |       [last_Record-Route:]
112 |       [last_From:]
113 |       [last_To:]
114 |       [last_Call-ID:]
115 |       [last_CSeq:]
116 |       Content-Length: 0
117 | 
118 |     ]]>
119 |   </send>
120 | 
121 |   <timewait milliseconds="2000"/><!-- retransmit 200's for a while -->
122 | 
123 | </scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->
124 | 


--------------------------------------------------------------------------------
/out-in/iinv-o200-iinv-o200-obye.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="iinv-o200-iinv-o200-obye.xml -- wjd 2015">
  4 | 
  5 |   <!--
  6 | 
  7 |   * Regular "wait for invite and send bye".
  8 |   * Except we expect a reinvite too.
  9 | 
 10 |   -->
 11 | 
 12 |   <recv request="INVITE" rrs="true"/>
 13 | 
 14 |   <send>
 15 |     <![CDATA[
 16 | 
 17 |       SIP/2.0 180 Ringing
 18 |       [last_Via:]
 19 |       [last_Record-Route:]
 20 |       [last_From:]
 21 |       [last_To:];tag=[pid]SIPpTag01[call_number]
 22 |       [last_Call-ID:]
 23 |       [last_CSeq:]
 24 |       Contact: sip:[service]@[local_ip]:[local_port]
 25 |       Content-Length: 0
 26 | 
 27 |     ]]>
 28 |   </send>
 29 | 
 30 |   <send retrans="500">
 31 |     <![CDATA[
 32 | 
 33 |       SIP/2.0 200 OK
 34 |       [last_Via:]
 35 |       [last_Record-Route:]
 36 |       [last_From:]
 37 |       [last_To:];tag=[pid]SIPpTag01[call_number]
 38 |       [last_Call-ID:]
 39 |       [last_CSeq:]
 40 |       Contact: sip:[service]@[local_ip]:[local_port]
 41 |       Content-Type: application/sdp
 42 |       Content-Length: [len]
 43 | 
 44 |       v=0
 45 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 46 |       s=-
 47 |       c=IN IP[media_ip_type] [media_ip]
 48 |       t=0 0
 49 |       m=audio [media_port] RTP/AVP 8 0
 50 |       a=rtpmap:8 PCMA/8000
 51 |       a=rtpmap:0 PCMU/8000
 52 | 
 53 |     ]]>
 54 |   </send>
 55 | 
 56 |   <recv request="ACK">
 57 |     <action>
 58 |       <ereg regexp=": *(.*)" search_in="hdr" header="To" assign_to="_,1"/><!-- with tag -->
 59 |       <ereg regexp=": *(.*)" search_in="hdr" header="From" assign_to="_,2"/>
 60 |     </action>
 61 |   </recv>
 62 |   <Reference variables="_"/>
 63 | 
 64 |   <nop>
 65 |     <action>
 66 |       <exec rtp_stream="beep_1sec_50x160b.alaw,1,8"/>
 67 |     </action>
 68 |   </nop>
 69 | 
 70 |   <!-- Wait for reINVITE because of transfer on other end. -->
 71 |   <recv request="INVITE" rrs="true"/>
 72 | 
 73 |   <!-- Reply with unchanged session -->
 74 |   <send retrans="500">
 75 |     <![CDATA[
 76 | 
 77 |       SIP/2.0 200 OK
 78 |       [last_Via:]
 79 |       [last_Record-Route:]
 80 |       [last_From:]
 81 |       [last_To:]
 82 |       [last_Call-ID:]
 83 |       [last_CSeq:]
 84 |       Contact: sip:[service]@[local_ip]:[local_port]
 85 |       Content-Type: application/sdp
 86 |       Content-Length: [len]
 87 | 
 88 |       v=0
 89 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 90 |       s=-
 91 |       c=IN IP[media_ip_type] [media_ip]
 92 |       t=0 0
 93 |       m=audio [media_port] RTP/AVP 8 0
 94 |       a=rtpmap:8 PCMA/8000
 95 |       a=rtpmap:0 PCMU/8000
 96 | 
 97 |     ]]>
 98 |   </send>
 99 | 
100 |   <recv request="ACK"/>
101 | 
102 |   <!-- Sleep a short while, keeping the new session open. -->
103 |   <nop>
104 |     <action>
105 |       <exec rtp_stream="beep_1sec_50x160b.alaw,1,8"/>
106 |     </action>
107 |   </nop>
108 |   <pause milliseconds="1000"/>
109 | 
110 |   <!-- We're done. Hang up. -->
111 |   <send start_txn="bye">
112 |     <![CDATA[
113 | 
114 |       BYE [next_url] SIP/2.0
115 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
116 |       [routes]
117 |       From: [$1]
118 |       To: [$2]
119 |       [last_Call-ID:]
120 |       CSeq: [cseq] BYE
121 |       Contact: sip:[service]@[local_ip]:[local_port]
122 |       Content-Length: 0
123 | 
124 |     ]]>
125 |   </send>
126 | 
127 |   <recv response="200" response_txn="bye"/>
128 | 
129 | </scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->
130 | 


--------------------------------------------------------------------------------
/out-in/iinv-o200-obye.xml:
--------------------------------------------------------------------------------
 1 | <?xml version="1.0" encoding="UTF-8" ?>
 2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
 3 | <scenario name="iinv-o200-obye.xml -- wjd 2014">
 4 | 
 5 |   <recv request="INVITE" rrs="true"/>
 6 | 
 7 |   <send>
 8 |     <![CDATA[
 9 | 
10 |       SIP/2.0 180 Ringing
11 |       [last_Via:]
12 |       [last_Record-Route:]
13 |       [last_From:]
14 |       [last_To:];tag=[pid]SIPpTag01[call_number]
15 |       [last_Call-ID:]
16 |       [last_CSeq:]
17 |       Contact: sip:[service]@[local_ip]:[local_port]
18 |       Content-Length: 0
19 | 
20 |     ]]>
21 |   </send>
22 | 
23 |   <send retrans="500">
24 |     <![CDATA[
25 | 
26 |       SIP/2.0 200 OK
27 |       [last_Via:]
28 |       [last_Record-Route:]
29 |       [last_From:]
30 |       [last_To:];tag=[pid]SIPpTag01[call_number]
31 |       [last_Call-ID:]
32 |       [last_CSeq:]
33 |       Contact: sip:[service]@[local_ip]:[local_port]
34 |       Content-Type: application/sdp
35 |       Content-Length: [len]
36 | 
37 |       v=0
38 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
39 |       s=-
40 |       c=IN IP[media_ip_type] [media_ip]
41 |       t=0 0
42 |       m=audio [media_port] RTP/AVP 8 0
43 |       a=rtpmap:8 PCMA/8000
44 |       a=rtpmap:0 PCMU/8000
45 | 
46 |     ]]>
47 |   </send>
48 | 
49 |   <recv request="ACK" rrs="true">
50 |     <action>
51 |       <ereg regexp=": *(.*)" search_in="hdr" header="To" assign_to="_,us"/><!-- with tag -->
52 |       <ereg regexp=": *(.*)" search_in="hdr" header="From" assign_to="_,them"/>
53 |     </action>
54 |   </recv>
55 |   <Reference variables="_"/>
56 | 
57 |   <!-- Send some audio to avoid rtptimeout timers kicking in. -->
58 |   <nop>
59 |     <action>
60 |       <exec rtp_stream="beep_1sec_50x160b.alaw,-1,8"/>
61 |     </action>
62 |   </nop>
63 | 
64 |   <pause milliseconds="1000"/>
65 | 
66 |   <send start_txn="bye" retrans="500">
67 |     <![CDATA[
68 | 
69 |       BYE [next_url] SIP/2.0
70 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
71 |       [routes]
72 |       From: [$us]
73 |       To: [$them]
74 |       [last_Call-ID:]
75 |       CSeq: [cseq] BYE
76 |       Contact: sip:[service]@[local_ip]:[local_port]
77 |       Content-Length: 0
78 | 
79 |     ]]>
80 |   </send>
81 | 
82 |   <recv response="200" response_txn="bye"/>
83 | 
84 | </scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->
85 | 


--------------------------------------------------------------------------------
/out-in/iinv-o200-oinv-i200-obye--w-cli-and-sessupd.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="iinv-o200-oinv-i200-obye--w-cli-and-sessupd.xml -- wjd 2015">
  4 | 
  5 |   <!-- Update the CLI.. first on 200, and then on reINVITE. -->
  6 | 
  7 |   <recv request="INVITE" rrs="true"/>
  8 | 
  9 |   <send>
 10 |     <![CDATA[
 11 | 
 12 |       SIP/2.0 180 Ringing
 13 |       [last_Via:]
 14 |       [last_Record-Route:]
 15 |       [last_From:]
 16 |       [last_To:];tag=[pid]SIPpTag01[call_number]
 17 |       [last_Call-ID:]
 18 |       [last_CSeq:]
 19 |       Contact: sip:[service]@[local_ip]:[local_port]
 20 |       Content-Length: 0
 21 |       Remote-Party-ID: "whatever" <sip:5432@[local_ip]>;privacy=off;screen=yes
 22 |       P-Asserted-Identity: "whatever" <sip:5432@[local_ip]>
 23 |       P-Preferred-Identity: "whatever" <sip:5432@[local_ip]>
 24 | 
 25 |     ]]>
 26 |   </send>
 27 | 
 28 |   <send retrans="500">
 29 |     <![CDATA[
 30 | 
 31 |       SIP/2.0 200 OK
 32 |       [last_Via:]
 33 |       [last_Record-Route:]
 34 |       [last_From:]
 35 |       [last_To:];tag=[pid]SIPpTag01[call_number]
 36 |       [last_Call-ID:]
 37 |       [last_CSeq:]
 38 |       Contact: sip:[service]@[local_ip]:[local_port]
 39 |       Supported: replaces
 40 |       Content-Type: application/sdp
 41 |       Content-Length: [len]
 42 |       Remote-Party-ID: "whatever" <sip:5432@[local_ip]>;privacy=off;screen=yes
 43 |       P-Asserted-Identity: "whatever" <sip:5432@[local_ip]>
 44 |       P-Preferred-Identity: "whatever" <sip:5432@[local_ip]>
 45 | 
 46 |       v=0
 47 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 48 |       s=-
 49 |       c=IN IP[media_ip_type] [media_ip]
 50 |       t=0 0
 51 |       m=audio [media_port] RTP/AVP 8 9
 52 |       a=rtpmap:8 PCMA/8000
 53 |       a=rtpmap:9 G722/8000
 54 | 
 55 |     ]]>
 56 |   </send>
 57 | 
 58 |   <recv request="ACK" rrs="true">
 59 |     <action>
 60 |       <ereg regexp=": *(.*)" search_in="hdr" header="To" assign_to="_,us"/><!-- with tag -->
 61 |       <ereg regexp=": *(.*)" search_in="hdr" header="From" assign_to="_,them"/>
 62 |     </action>
 63 |   </recv>
 64 |   <Reference variables="_"/>
 65 | 
 66 |   <nop>
 67 |     <action>
 68 |       <exec rtp_stream="beep_1sec_50x160b.alaw,1,8"/>
 69 |     </action>
 70 |   </nop>
 71 |   <pause milliseconds="1000"/>
 72 | 
 73 |   <!-- Update RPID, try only G722 codec. -->
 74 |   <send retrans="500" start_txn="invite">
 75 |     <![CDATA[
 76 | 
 77 |       INVITE [next_url] SIP/2.0
 78 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 79 |       [routes]
 80 |       From: [$us]
 81 |       To: [$them]
 82 |       Call-ID: [call_id]
 83 |       CSeq: [cseq] INVITE
 84 |       Contact: sip:[service]@[local_ip]:[local_port]
 85 |       Content-Type: application/sdp
 86 |       Content-Length: [len]
 87 |       Remote-Party-ID: "whatever2" <sip:54321@[local_ip]>;privacy=off;screen=yes
 88 |       P-Asserted-Identity: "whatever2" <sip:54321@[local_ip]>
 89 |       P-Preferred-Identity: "whatever2" <sip:54321@[local_ip]>
 90 | 
 91 |       v=0
 92 |       o=user1 53655765 2353687638 IN IP[local_ip_type] [local_ip]
 93 |       s=-
 94 |       c=IN IP[media_ip_type] [media_ip]
 95 |       t=0 0
 96 |       m=audio [media_port] RTP/AVP 9
 97 |       a=rtpmap:9 G722/8000
 98 | 
 99 |     ]]>
100 |   </send>
101 | 
102 |   <recv response="100" optional="true" response_txn="invite"/>
103 | 
104 |   <recv response="200" rrs="true" response_txn="invite"/>
105 | 
106 |   <send ack_txn="invite">
107 |     <![CDATA[
108 | 
109 |       ACK [next_url] SIP/2.0
110 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
111 |       [routes]
112 |       From: [$us]
113 |       To: [$them]
114 |       Call-ID: [call_id]
115 |       CSeq: [cseq] ACK
116 |       Contact: sip:[service]@[local_ip]:[local_port]
117 |       Content-Length: 0
118 | 
119 |     ]]>
120 |   </send>
121 | 
122 |   <nop>
123 |     <action>
124 |       <exec rtp_stream="beep_1sec_50x160b.alaw,1,8"/><!-- pretend G722, send A-law anyway -->
125 |     </action>
126 |   </nop>
127 |   <pause milliseconds="1000"/>
128 | 
129 |   <send start_txn="bye">
130 |     <![CDATA[
131 | 
132 |       BYE [next_url] SIP/2.0
133 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
134 |       [routes]
135 |       From: [$us]
136 |       To: [$them]
137 |       [last_Call-ID:]
138 |       CSeq: [cseq] BYE
139 |       Contact: sip:[service]@[local_ip]:[local_port]
140 |       Content-Length: 0
141 | 
142 |     ]]>
143 |   </send>
144 | 
145 |   <recv response="200" response_txn="bye"/>
146 | 
147 | </scenario>
148 | <!-- vim: set ts=8 sw=2 sts=2 et ai: -->
149 | 


--------------------------------------------------------------------------------
/out-in/iinv-o200-oinv-i200-oi3pcc-orefer-obye.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="iinv-o200-oinv-i200-oi3pcc-orefer-obye.xml -- wjd 2015">
  4 | 
  5 |   <!--
  6 | 
  7 |   * Wait for invite.
  8 |   * Initiate hold (a=sendonly).
  9 |   * Poke 3pcc to start setting up a new leg.
 10 |   * Get other dialog info from 3pcc.
 11 |   * Refer other end of this call to the other dialog.
 12 |   * Hang up this leg.
 13 | 
 14 |   -->
 15 | 
 16 |   <recv request="INVITE" rrs="true"/>
 17 | 
 18 |   <send>
 19 |     <![CDATA[
 20 | 
 21 |       SIP/2.0 180 Ringing
 22 |       [last_Via:]
 23 |       [last_Record-Route:]
 24 |       [last_From:]
 25 |       [last_To:];tag=[pid]SIPpTag01[call_number]
 26 |       [last_Call-ID:]
 27 |       [last_CSeq:]
 28 |       Contact: sip:[service]@[local_ip]:[local_port]
 29 |       Content-Length: 0
 30 | 
 31 |     ]]>
 32 |   </send>
 33 | 
 34 |   <send retrans="500">
 35 |     <![CDATA[
 36 | 
 37 |       SIP/2.0 200 OK
 38 |       [last_Via:]
 39 |       [last_Record-Route:]
 40 |       [last_From:]
 41 |       [last_To:];tag=[pid]SIPpTag01[call_number]
 42 |       [last_Call-ID:]
 43 |       [last_CSeq:]
 44 |       Contact: sip:[service]@[local_ip]:[local_port]
 45 |       Content-Type: application/sdp
 46 |       Content-Length: [len]
 47 | 
 48 |       v=0
 49 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 50 |       s=-
 51 |       c=IN IP[media_ip_type] [media_ip]
 52 |       t=0 0
 53 |       m=audio [media_port] RTP/AVP 8 0
 54 |       a=rtpmap:8 PCMA/8000
 55 |       a=rtpmap:0 PCMU/8000
 56 | 
 57 |     ]]>
 58 |   </send>
 59 | 
 60 |   <recv request="ACK" rrs="true">
 61 |     <action>
 62 |       <ereg regexp=": *(.*)" search_in="hdr" header="To" assign_to="_,us"/><!-- with tag -->
 63 |       <ereg regexp=": *(.*)" search_in="hdr" header="From" assign_to="_,them"/>
 64 |     </action>
 65 |   </recv>
 66 |   <Reference variables="_"/>
 67 | 
 68 |   <nop>
 69 |     <action>
 70 |       <exec rtp_stream="beep_1sec_50x160b.alaw,1,8"/>
 71 |     </action>
 72 |   </nop>
 73 |   <pause milliseconds="1000"/>
 74 | 
 75 |   <!-- Initiate hold (a=sendonly) -->
 76 |   <send retrans="500" start_txn="invite">
 77 |     <![CDATA[
 78 | 
 79 |       INVITE [next_url] SIP/2.0
 80 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 81 |       [routes]
 82 |       From: [$us]
 83 |       To: [$them]
 84 |       Call-ID: [call_id]
 85 |       CSeq: [cseq] INVITE
 86 |       Contact: sip:[service]@[local_ip]:[local_port]
 87 |       Content-Type: application/sdp
 88 |       Content-Length: [len]
 89 | 
 90 |       v=0
 91 |       o=user1 53655765 2353687638 IN IP[local_ip_type] [local_ip]
 92 |       s=-
 93 |       c=IN IP[media_ip_type] 0.0.0.0
 94 |       t=0 0
 95 |       m=audio [media_port] RTP/AVP 8 0
 96 |       a=rtpmap:8 PCMA/8000
 97 |       a=rtpmap:0 PCMU/8000
 98 |       a=sendonly
 99 | 
100 |     ]]>
101 |   </send>
102 | 
103 |   <recv response="100" optional="true" response_txn="invite"/>
104 | 
105 |   <recv response="200" rrs="true" response_txn="invite"/>
106 | 
107 |   <send ack_txn="invite">
108 |     <![CDATA[
109 | 
110 |       ACK [next_url] SIP/2.0
111 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
112 |       [routes]
113 |       From: [$us]
114 |       To: [$them]
115 |       Call-ID: [call_id]
116 |       CSeq: [cseq] ACK
117 |       Contact: sip:[service]@[local_ip]:[local_port]
118 |       Content-Length: 0
119 | 
120 |     ]]>
121 |   </send>
122 | 
123 |   <!-- Peer is placed on hold. Poke 3pcc to initiate a second dialog.
124 |        (Call-ID must be in there. It's a 3pcc limitation/feature.) -->
125 |   <sendCmd>
126 |     <![CDATA[
127 | 
128 |       Call-ID: [call_id]
129 | 
130 |       Poke the peer. (And tell them our Call-ID at the same time.)
131 | 
132 |     ]]>
133 |   </sendCmd>
134 | 
135 |   <!-- Wait for 3pcc to come up with dialog info that we can steal. -->
136 |   <recvCmd>
137 |     <action>
138 |       <!-- Store next_url in $3 -->
139 |       <ereg regexp="(sip:[^@]+@[^>;]+)"
140 |             search_in="hdr"
141 |             header="Contact:"
142 |             check_it="true"
143 |             assign_to="3"/>
144 |       <!-- Store last_From-tag in $4 -->
145 |       <ereg regexp="tag=([^;]*)"
146 |             search_in="hdr"
147 |             header="From:"
148 |             check_it="true"
149 |             assign_to="9,4"/>
150 |       <!-- Store last_To-tag in $5 -->
151 |       <ereg regexp="tag=([^;]*)"
152 |             search_in="hdr"
153 |             header="To:"
154 |             check_it="true"
155 |             assign_to="9,5"/>
156 |     </action>
157 |   </recvCmd>
158 | 
159 |   <!-- Refer the new dialog to the old (our) dialog.
160 |        Note how we "dedup///" the call_id this time: the peer
161 |        uses "dedup///[call_id]" as their call id.  -->
162 |   <send>
163 |     <![CDATA[
164 | 
165 |       REFER [next_url] SIP/2.0
166 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
167 |       [routes]
168 |       From: [$us]
169 |       To: [$them]
170 |       Call-ID: [call_id]
171 |       CSeq: [cseq] REFER
172 |       Contact: sip:[service]@[local_ip]:[local_port]
173 |       Referred-By: sip:[service]@[local_ip]:[local_port]
174 |       Refer-To: <[$3]?Replaces=dedup///[call_id]%3Bto-tag%3D[$5]%3Bfrom-tag%3D[$4]>
175 |       Content-Length: 0
176 | 
177 |   ]]>
178 |   </send>
179 | 
180 |   <!-- Sometimes the 202 comes before the notify, sometimes after.
181 |        We'll allow an optional NOTIFY first, and wait for the retransmit to
182 |        answer it. -->
183 |   <recv request="NOTIFY" optional="true"/>
184 |   <recv response="202"/>
185 | 
186 |   <!-- This should contain a 200 success message, or 180 ringing,
187 |        depending on the other scenarios. -->
188 |   <recv request="NOTIFY"/>
189 | 
190 |   <send>
191 |     <![CDATA[
192 | 
193 |       SIP/2.0 200 OK
194 |       [last_Via:]
195 |       [last_From:]
196 |       [last_To:]
197 |       [last_Call-ID:]
198 |       [last_CSeq:]
199 |       [last_Event:]
200 |       Contact: sip:[service]@[local_ip]:[local_port]
201 |       Content-Length: 0
202 | 
203 |     ]]>
204 |   </send>
205 | 
206 |   <!-- Hang up this old leg. -->
207 |   <send start_txn="bye">
208 |     <![CDATA[
209 | 
210 |       BYE [next_url] SIP/2.0
211 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
212 |       [routes]
213 |       From: [$us]
214 |       To: [$them]
215 |       [last_Call-ID:]
216 |       CSeq: [cseq] BYE
217 |       Contact: sip:[service]@[local_ip]:[local_port]
218 |       Content-Length: 0
219 | 
220 |     ]]>
221 |   </send>
222 | 
223 |   <recv response="200" response_txn="bye"/>
224 | 
225 | </scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->
226 | 


--------------------------------------------------------------------------------
/out-in/iinv-o200-oinv-i200-oinv-i200-wait-obye.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="iinv-o200-oinv-i200-oinv-i200-wait-obye.xml -- wjd 2015">
  4 | 
  5 |   <!--
  6 | 
  7 |   * Wait for invite (answer with supported without timers).
  8 |   * Initiate hold (a=sendonly, no supported-header).
  9 |   * Initiate unhold (a=sendrecv, no supported-header).
 10 |   * Wait for more than 90 seconds. This could trigger a bug in Asterisk
 11 |     version 11.20 (and later, and earlier) where chan_sip re-enables
 12 |     session timers after reinvite, even though they should not be
 13 |     enabled.
 14 |   * Hang up this leg.
 15 | 
 16 |   -->
 17 | 
 18 |   <recv request="INVITE" rrs="true"/>
 19 | 
 20 |   <send>
 21 |     <![CDATA[
 22 | 
 23 |       SIP/2.0 180 Ringing
 24 |       [last_Via:]
 25 |       [last_Record-Route:]
 26 |       [last_From:]
 27 |       [last_To:];tag=[pid]SIPpTag01[call_number]
 28 |       [last_Call-ID:]
 29 |       [last_CSeq:]
 30 |       Contact: sip:[service]@[local_ip]:[local_port]
 31 |       Content-Length: 0
 32 | 
 33 |     ]]>
 34 |   </send>
 35 | 
 36 |   <send retrans="500">
 37 |     <![CDATA[
 38 | 
 39 |       SIP/2.0 200 OK
 40 |       [last_Via:]
 41 |       [last_Record-Route:]
 42 |       [last_From:]
 43 |       [last_To:];tag=[pid]SIPpTag01[call_number]
 44 |       [last_Call-ID:]
 45 |       [last_CSeq:]
 46 |       Contact: sip:[service]@[local_ip]:[local_port]
 47 |       Supported: replaces
 48 |       Content-Type: application/sdp
 49 |       Content-Length: [len]
 50 | 
 51 |       v=0
 52 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 53 |       s=-
 54 |       c=IN IP[media_ip_type] [media_ip]
 55 |       t=0 0
 56 |       m=audio [media_port] RTP/AVP 8 0
 57 |       a=rtpmap:8 PCMA/8000
 58 |       a=rtpmap:0 PCMU/8000
 59 | 
 60 |     ]]>
 61 |   </send>
 62 | 
 63 |   <recv request="ACK" rrs="true">
 64 |     <action>
 65 |       <ereg regexp=": *(.*)" search_in="hdr" header="To" assign_to="_,us"/><!-- with tag -->
 66 |       <ereg regexp=": *(.*)" search_in="hdr" header="From" assign_to="_,them"/>
 67 |     </action>
 68 |   </recv>
 69 |   <Reference variables="_"/>
 70 | 
 71 |   <nop>
 72 |     <action>
 73 |       <exec rtp_stream="beep_1sec_50x160b.alaw,1,8"/>
 74 |     </action>
 75 |   </nop>
 76 |   <pause milliseconds="1000"/>
 77 | 
 78 |   <!-- Initiate hold (a=sendonly) -->
 79 |   <send retrans="500" start_txn="invite">
 80 |     <![CDATA[
 81 | 
 82 |       INVITE [next_url] SIP/2.0
 83 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 84 |       [routes]
 85 |       From: [$us]
 86 |       To: [$them]
 87 |       Call-ID: [call_id]
 88 |       CSeq: [cseq] INVITE
 89 |       Contact: sip:[service]@[local_ip]:[local_port]
 90 |       Content-Type: application/sdp
 91 |       Content-Length: [len]
 92 | 
 93 |       v=0
 94 |       o=user1 53655765 2353687638 IN IP[local_ip_type] [local_ip]
 95 |       s=-
 96 |       c=IN IP[media_ip_type] 0.0.0.0
 97 |       t=0 0
 98 |       m=audio [media_port] RTP/AVP 8 0
 99 |       a=rtpmap:8 PCMA/8000
100 |       a=rtpmap:0 PCMU/8000
101 |       a=sendonly
102 | 
103 |     ]]>
104 |   </send>
105 | 
106 |   <recv response="100" optional="true" response_txn="invite"/>
107 | 
108 |   <recv response="200" rrs="true" response_txn="invite"/>
109 | 
110 |   <send ack_txn="invite">
111 |     <![CDATA[
112 | 
113 |       ACK [next_url] SIP/2.0
114 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
115 |       [routes]
116 |       From: [$us]
117 |       To: [$them]
118 |       Call-ID: [call_id]
119 |       CSeq: [cseq] ACK
120 |       Contact: sip:[service]@[local_ip]:[local_port]
121 |       Content-Length: 0
122 | 
123 |     ]]>
124 |   </send>
125 | 
126 |   <nop>
127 |     <action>
128 |       <exec rtp_stream="beep_1sec_50x160b.alaw,1,8"/>
129 |     </action>
130 |   </nop>
131 |   <pause milliseconds="1000"/>
132 | 
133 |   <!-- Initiate unhold (a=sendrecv) -->
134 |   <send retrans="500" start_txn="invite">
135 |     <![CDATA[
136 | 
137 |       INVITE [next_url] SIP/2.0
138 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
139 |       [routes]
140 |       From: [$us]
141 |       To: [$them]
142 |       Call-ID: [call_id]
143 |       CSeq: [cseq] INVITE
144 |       Contact: sip:[service]@[local_ip]:[local_port]
145 |       Content-Type: application/sdp
146 |       Content-Length: [len]
147 | 
148 |       v=0
149 |       o=user1 53655765 2353687639 IN IP[local_ip_type] [local_ip]
150 |       s=-
151 |       c=IN IP[media_ip_type] [media_ip]
152 |       t=0 0
153 |       m=audio [media_port] RTP/AVP 8 0
154 |       a=rtpmap:8 PCMA/8000
155 |       a=rtpmap:0 PCMU/8000
156 |       a=sendrecv
157 | 
158 |     ]]>
159 |   </send>
160 | 
161 |   <recv response="100" optional="true" response_txn="invite"/>
162 | 
163 |   <recv response="200" rrs="true" response_txn="invite"/>
164 | 
165 |   <send ack_txn="invite">
166 |     <![CDATA[
167 | 
168 |       ACK [next_url] SIP/2.0
169 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
170 |       [routes]
171 |       From: [$us]
172 |       To: [$them]
173 |       Call-ID: [call_id]
174 |       CSeq: [cseq] ACK
175 |       Contact: sip:[service]@[local_ip]:[local_port]
176 |       Content-Length: 0
177 | 
178 |     ]]>
179 |   </send>
180 | 
181 |   <!-- Time to sit and wait. Add some playback so we don't get cancelled
182 |        by the rtptimeout. -->
183 |   <nop>
184 |     <action>
185 |       <exec rtp_stream="beep_1sec_50x160b.alaw,-1,8"/>
186 |     </action>
187 |   </nop>
188 |   <pause milliseconds="100000"/>
189 | 
190 |   <!-- Hang up this old leg. -->
191 |   <send start_txn="bye">
192 |     <![CDATA[
193 | 
194 |       BYE [next_url] SIP/2.0
195 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
196 |       [routes]
197 |       From: [$us]
198 |       To: [$them]
199 |       [last_Call-ID:]
200 |       CSeq: [cseq] BYE
201 |       Contact: sip:[service]@[local_ip]:[local_port]
202 |       Content-Length: 0
203 | 
204 |     ]]>
205 |   </send>
206 | 
207 |   <recv response="200" response_txn="bye"/>
208 | 
209 | </scenario>
210 | <!-- vim: set ts=8 sw=2 sts=2 et ai: -->
211 | 


--------------------------------------------------------------------------------
/out-in/iinv-o302.xml:
--------------------------------------------------------------------------------
 1 | <?xml version="1.0" encoding="UTF-8" ?>
 2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
 3 | <scenario name="iinv-o302.xml -- wjd 2014">
 4 |   <recv request="INVITE" rrs="true"/>
 5 | 
 6 |   <send>
 7 |     <![CDATA[
 8 | 
 9 |       SIP/2.0 100 Trying
10 |       [last_Via:]
11 |       [last_Record-Route:]
12 |       [last_From:]
13 |       [last_To:]
14 |       [last_Call-ID:]
15 |       [last_CSeq:]
16 |       Content-Length: 0
17 | 
18 |     ]]>
19 |   </send>
20 | 
21 |   <pause milliseconds="1000"/>
22 | 
23 |   <send retrans="500">
24 |     <![CDATA[
25 | 
26 |       SIP/2.0 302 OK
27 |       [last_Via:]
28 |       [last_Record-Route:]
29 |       [last_From:]
30 |       [last_To:];tag=[pid]SIPpTag01[call_number]
31 |       Contact: <[contact]>
32 |       [last_Call-ID:]
33 |       [last_CSeq:]
34 |       Content-Length: 0
35 | 
36 |     ]]>
37 |   </send>
38 | 
39 |   <recv request="ACK"/>
40 | 
41 |   <!-- Keep call open to be able to retransmit 486 -->
42 |   <timewait milliseconds="2000"/>
43 | </scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->
44 | 


--------------------------------------------------------------------------------
/out-in/iinv-o486.xml:
--------------------------------------------------------------------------------
 1 | <?xml version="1.0" encoding="UTF-8" ?>
 2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
 3 | <scenario name="iinv-o486.xml -- wjd 2014">
 4 |   <recv request="INVITE" rrs="true"/>
 5 | 
 6 |   <send>
 7 |     <![CDATA[
 8 | 
 9 |       SIP/2.0 100 Trying
10 |       [last_Via:]
11 |       [last_Record-Route:]
12 |       [last_From:]
13 |       [last_To:]
14 |       [last_Call-ID:]
15 |       [last_CSeq:]
16 |       Content-Length: 0
17 | 
18 |     ]]>
19 |   </send>
20 | 
21 |   <pause milliseconds="1000"/>
22 | 
23 |   <send retrans="500">
24 |     <![CDATA[
25 | 
26 |       SIP/2.0 486 OK
27 |       [last_Via:]
28 |       [last_Record-Route:]
29 |       [last_From:]
30 |       [last_To:];tag=[pid]SIPpTag01[call_number]
31 |       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
32 |       [last_Call-ID:]
33 |       [last_CSeq:]
34 |       Content-Length: 0
35 | 
36 |     ]]>
37 |   </send>
38 | 
39 |   <recv request="ACK"/>
40 | 
41 |   <!-- Keep call open to be able to retransmit 486 -->
42 |   <timewait milliseconds="2000"/>
43 | </scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->
44 | 


--------------------------------------------------------------------------------
/out-in/oinv-i180-ocancel.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="oinv-i180-ocancel.xml -- wjd 2014">
  4 | 
  5 |   <send retrans="500" start_txn="invite">
  6 |     <![CDATA[
  7 | 
  8 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
  9 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 10 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 11 |       To: sip:[tel]@[remote_ip]:[remote_port]
 12 |       Call-ID: [call_id]
 13 |       CSeq: [cseq] INVITE
 14 |       Contact: sip:[service]@[local_ip]:[local_port]
 15 |       Content-Type: application/sdp
 16 |       Content-Length: [len]
 17 | 
 18 |       v=0
 19 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 20 |       s=-
 21 |       c=IN IP[media_ip_type] [media_ip]
 22 |       t=0 0
 23 |       m=audio [media_port] RTP/AVP 8 0
 24 |       a=rtpmap:8 PCMA/8000
 25 |       a=rtpmap:0 PCMU/8000
 26 | 
 27 |     ]]>
 28 |   </send>
 29 | 
 30 |   <recv response="100" optional="true" response_txn="invite"/>
 31 | 
 32 |   <recv response="401" optional="true" auth="true" next="invite-with-auth" response_txn="invite">
 33 |     <action>
 34 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 35 |     </action>
 36 |   </recv>
 37 | 
 38 |   <recv response="407" auth="true" next="invite-with-auth" response_txn="invite">
 39 |     <action>
 40 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 41 |     </action>
 42 |   </recv>
 43 | 
 44 |   <label id="invite-with-auth"/>
 45 | 
 46 |   <!-- ACK is within transaction because of non-2xx. Reuse Via branch.      -->
 47 |   <!-- We don't use last_Via: because it would include rport stuff from     -->
 48 |   <!-- remote.                                                              -->
 49 |   <send ack_txn="invite">
 50 |     <![CDATA[
 51 | 
 52 |       ACK sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 53 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$1]
 54 |       [last_From:]
 55 |       [last_To:]
 56 |       [last_Call-ID:]
 57 |       CSeq: [cseq] ACK
 58 |       Contact: sip:[service]@[local_ip]:[local_port]
 59 |       Content-Length: 0
 60 | 
 61 |     ]]>
 62 |   </send>
 63 | 
 64 |   <send retrans="500" start_txn="invite">
 65 |     <![CDATA[
 66 | 
 67 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 68 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 69 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 70 |       To: sip:[tel]@[remote_ip]:[remote_port]
 71 |       [authentication]
 72 |       Call-ID: [call_id]
 73 |       CSeq: [cseq] INVITE
 74 |       Contact: sip:[service]@[local_ip]:[local_port]
 75 |       Content-Type: application/sdp
 76 |       Content-Length: [len]
 77 | 
 78 |       v=0
 79 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 80 |       s=-
 81 |       c=IN IP[media_ip_type] [media_ip]
 82 |       t=0 0
 83 |       m=audio [media_port] RTP/AVP 8 0
 84 |       a=rtpmap:8 PCMA/8000
 85 |       a=rtpmap:0 PCMU/8000
 86 | 
 87 |     ]]>
 88 |   </send>
 89 | 
 90 |   <recv response="100" response_txn="invite">
 91 |     <action>
 92 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 93 |     </action>
 94 |   </recv>
 95 | 
 96 |   <recv response="180" response_txn="invite"/>
 97 | 
 98 |   <pause milliseconds="2000"/>
 99 | 
100 |   <send start_txn="cancel">
101 |     <![CDATA[
102 | 
103 |       CANCEL sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
104 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$1]
105 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
106 |       To: sip:[tel]@[remote_ip]:[remote_port]
107 |       Call-ID: [call_id]
108 |       CSeq: [cseq] CANCEL
109 |       Max-Forwards: 30
110 | 
111 |     ]]>
112 |   </send>
113 | 
114 |   <recv response="200" response_txn="cancel"/>
115 | 
116 |   <recv response="487" response_txn="invite"/>
117 | 
118 |   <send ack_txn="invite">
119 |     <![CDATA[
120 | 
121 |       ACK sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
122 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$1]
123 |       [last_From:]
124 |       [last_To:]
125 |       [last_Call-ID:]
126 |       CSeq: [cseq] ACK
127 |       Max-Forwards: 30
128 |       Content-Length: 0
129 | 
130 |     ]]>
131 |   </send>
132 | 
133 |   <timewait milliseconds="1000"/>
134 | 
135 | </scenario>
136 | 


--------------------------------------------------------------------------------
/out-in/oinv-i200-ibye--w-rpid.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="oinv-i200-ibye--w-rpid.xml -- wjd 2014-2015">
  4 | 
  5 |   <!--
  6 | 
  7 |   This call setup is used for internal as well as external (inward)
  8 |   calls. So it requires the possibility to do unauthenticated calls.
  9 | 
 10 |   -->
 11 | 
 12 |   <send retrans="500" start_txn="invite">
 13 |     <![CDATA[
 14 | 
 15 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 16 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 17 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 18 |       To: sip:[tel]@[remote_ip]:[remote_port]
 19 |       Call-ID: [call_id]
 20 |       CSeq: [cseq] INVITE
 21 |       Contact: sip:[service]@[local_ip]:[local_port]
 22 |       Remote-Party-ID: [rpid]
 23 |       Content-Type: application/sdp
 24 |       Content-Length: [len]
 25 | 
 26 |       v=0
 27 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 28 |       s=-
 29 |       c=IN IP[media_ip_type] [media_ip]
 30 |       t=0 0
 31 |       m=audio [media_port] RTP/AVP 8 0
 32 |       a=rtpmap:8 PCMA/8000
 33 |       a=rtpmap:0 PCMU/8000
 34 | 
 35 |     ]]>
 36 |   </send>
 37 | 
 38 |   <recv response="100" optional="true" response_txn="invite"/>
 39 | 
 40 |   <recv response="401" optional="true" auth="true" next="invite-with-auth" response_txn="invite">
 41 |     <action>
 42 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 43 |     </action>
 44 |   </recv>
 45 | 
 46 |   <recv response="407" optional="true" auth="true" next="invite-with-auth" response_txn="invite">
 47 |     <action>
 48 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 49 |     </action>
 50 |   </recv>
 51 | 
 52 |   <recv response="180" optional="true" response_txn="invite"/>
 53 | 
 54 |   <recv response="183" optional="true" response_txn="invite"/>
 55 | 
 56 |   <recv response="200" next="invite-ack" rrs="true" response_txn="invite"/>
 57 | 
 58 |   <label id="invite-with-auth"/>
 59 | 
 60 |   <!-- ACK is within transaction because of non-2xx. Reuse Via branch.      -->
 61 |   <!-- We don't use last_Via: because it would include rport stuff from     -->
 62 |   <!-- remote.                                                              -->
 63 |   <send ack_txn="invite">
 64 |     <![CDATA[
 65 | 
 66 |       ACK sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 67 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$1]
 68 |       [last_From:]
 69 |       [last_To:]
 70 |       [last_Call-ID:]
 71 |       CSeq: [cseq] ACK
 72 |       Contact: sip:[service]@[local_ip]:[local_port]
 73 |       Content-Length: 0
 74 | 
 75 |     ]]>
 76 |   </send>
 77 | 
 78 |   <send retrans="500" start_txn="invite">
 79 |     <![CDATA[
 80 | 
 81 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 82 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 83 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 84 |       To: sip:[tel]@[remote_ip]:[remote_port]
 85 |       [authentication]
 86 |       Call-ID: [call_id]
 87 |       CSeq: [cseq] INVITE
 88 |       Contact: sip:[service]@[local_ip]:[local_port]
 89 |       Remote-Party-ID: [rpid]
 90 |       Content-Type: application/sdp
 91 |       Content-Length: [len]
 92 | 
 93 |       v=0
 94 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 95 |       s=-
 96 |       c=IN IP[media_ip_type] [media_ip]
 97 |       t=0 0
 98 |       m=audio [media_port] RTP/AVP 8 0
 99 |       a=rtpmap:8 PCMA/8000
100 |       a=rtpmap:0 PCMU/8000
101 | 
102 |     ]]>
103 |   </send>
104 | 
105 |   <recv response="100" response_txn="invite"/>
106 | 
107 |   <recv response="180" optional="true" response_txn="invite"/>
108 | 
109 |   <recv response="183" optional="true" response_txn="invite"/>
110 | 
111 |   <recv response="200" rrs="true" response_txn="invite"/>
112 | 
113 |   <label id="invite-ack"/>
114 | 
115 |   <send ack_txn="invite">
116 |     <![CDATA[
117 | 
118 |       ACK [next_url] SIP/2.0
119 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
120 |       [routes]
121 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
122 |       To: sip:[tel]@[remote_ip]:[remote_port][peer_tag_param]
123 |       Call-ID: [call_id]
124 |       CSeq: [cseq] ACK
125 |       Contact: sip:[service]@[local_ip]:[local_port]
126 |       Content-Length: 0
127 | 
128 |     ]]>
129 |   </send>
130 | 
131 |   <nop>
132 |     <action>
133 |       <exec rtp_stream="beep_1sec_50x160b.alaw,1,8"/>
134 |     </action>
135 |   </nop>
136 | 
137 |   <recv request="BYE"/>
138 | 
139 |   <send>
140 |     <![CDATA[
141 | 
142 |       SIP/2.0 200 OK
143 |       [last_Via:]
144 |       [last_Record-Route:]
145 |       [last_From:]
146 |       [last_To:]
147 |       [last_Call-ID:]
148 |       [last_CSeq:]
149 |       Content-Length: 0
150 | 
151 |     ]]>
152 |   </send>
153 | 
154 |   <timewait milliseconds="2000"/><!-- retransmit 200's for a while -->
155 | 
156 | </scenario>
157 | 


--------------------------------------------------------------------------------
/out-in/oinv-i200-ibye--w-sesstimer-uac.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="oinv-i200-ibye--w-sesstimer-uac.xml -- wjd 2015">
  4 | 
  5 |   <!--
  6 | 
  7 |   * Dial out with short Session-Expires header.
  8 |     (The peer will expect a refresher reinvite from us in about 91-30
  9 |     seconds.)
 10 |   * Wait for BYE.
 11 | 
 12 |   -->
 13 | 
 14 |   <send retrans="500" start_txn="invite">
 15 |     <![CDATA[
 16 | 
 17 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 18 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 19 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 20 |       To: sip:[tel]@[remote_ip]:[remote_port]
 21 |       Call-ID: [call_id]
 22 |       CSeq: [cseq] INVITE
 23 |       Contact: sip:[service]@[local_ip]:[local_port]
 24 |       Supports: replaces, timer
 25 |       Require: timer
 26 |       Session-Expires: 91;refresher=uac
 27 |       Content-Type: application/sdp
 28 |       Content-Length: [len]
 29 | 
 30 |       v=0
 31 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 32 |       s=-
 33 |       c=IN IP[media_ip_type] [media_ip]
 34 |       t=0 0
 35 |       m=audio [media_port] RTP/AVP 8 0
 36 |       a=rtpmap:8 PCMA/8000
 37 |       a=rtpmap:0 PCMU/8000
 38 | 
 39 |     ]]>
 40 |   </send>
 41 | 
 42 |   <recv response="100" optional="true" response_txn="invite"/>
 43 | 
 44 |   <recv response="401" optional="true" auth="true" next="invite-with-auth" response_txn="invite">
 45 |     <action>
 46 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="viabranch"/>
 47 |     </action>
 48 |   </recv>
 49 | 
 50 |   <recv response="407" auth="true" next="invite-with-auth" response_txn="invite">
 51 |     <action>
 52 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="viabranch"/>
 53 |     </action>
 54 |   </recv>
 55 | 
 56 |   <label id="invite-with-auth"/>
 57 | 
 58 |   <!-- ACK is within transaction because of non-2xx. Reuse Via branch.      -->
 59 |   <!-- We don't use last_Via: because it would include rport stuff from     -->
 60 |   <!-- remote.                                                              -->
 61 |   <send ack_txn="invite">
 62 |     <![CDATA[
 63 | 
 64 |       ACK sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 65 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$viabranch]
 66 |       [last_From:]
 67 |       [last_To:]
 68 |       [last_Call-ID:]
 69 |       CSeq: [cseq] ACK
 70 |       Contact: sip:[service]@[local_ip]:[local_port]
 71 |       Content-Length: 0
 72 | 
 73 |     ]]>
 74 |   </send>
 75 | 
 76 |   <send retrans="500" start_txn="invite">
 77 |     <![CDATA[
 78 | 
 79 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 80 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 81 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 82 |       To: sip:[tel]@[remote_ip]:[remote_port]
 83 |       [authentication]
 84 |       Call-ID: [call_id]
 85 |       CSeq: [cseq] INVITE
 86 |       Contact: sip:[service]@[local_ip]:[local_port]
 87 |       Supports: replaces, timer
 88 |       Require: timer
 89 |       Session-Expires: 91;refresher=uac
 90 |       Content-Type: application/sdp
 91 |       Content-Length: [len]
 92 | 
 93 |       v=0
 94 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 95 |       s=-
 96 |       c=IN IP[media_ip_type] [media_ip]
 97 |       t=0 0
 98 |       m=audio [media_port] RTP/AVP 8 0
 99 |       a=rtpmap:8 PCMA/8000
100 |       a=rtpmap:0 PCMU/8000
101 | 
102 |     ]]>
103 |   </send>
104 | 
105 |   <recv response="100" response_txn="invite"/>
106 | 
107 |   <recv response="180" optional="true" response_txn="invite"/>
108 | 
109 |   <recv response="183" optional="true" response_txn="invite"/>
110 | 
111 |   <recv response="200" rrs="true" response_txn="invite"/>
112 | 
113 |   <send ack_txn="invite">
114 |     <![CDATA[
115 | 
116 |       ACK [next_url] SIP/2.0
117 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
118 |       [routes]
119 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
120 |       To: sip:[tel]@[remote_ip]:[remote_port][peer_tag_param]
121 |       Call-ID: [call_id]
122 |       CSeq: [cseq] ACK
123 |       Contact: sip:[service]@[local_ip]:[local_port]
124 |       Content-Length: 0
125 | 
126 |     ]]>
127 |   </send>
128 | 
129 |   <!-- Send some audio to avoid rtptimeout timers kicking in. -->
130 |   <nop>
131 |     <action>
132 |       <exec rtp_stream="beep_1sec_50x160b.alaw,-1,8"/>
133 |     </action>
134 |   </nop>
135 | 
136 |   <recv request="BYE"/>
137 | 
138 |   <send>
139 |     <![CDATA[
140 | 
141 |       SIP/2.0 200 OK
142 |       [last_Via:]
143 |       [last_Record-Route:]
144 |       [last_From:]
145 |       [last_To:]
146 |       [last_Call-ID:]
147 |       [last_CSeq:]
148 |       Content-Length: 0
149 | 
150 |     ]]>
151 |   </send>
152 | 
153 |   <timewait milliseconds="2000"/><!-- retransmit 200's for a while -->
154 | 
155 | </scenario>
156 | <!-- vim: set ts=8 sw=2 sts=2 et ai: -->
157 | 


--------------------------------------------------------------------------------
/out-in/oinv-i200-ibye.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="oinv-i200-ibye.xml -- wjd 2014">
  4 | 
  5 |   <send retrans="500" start_txn="invite">
  6 |     <![CDATA[
  7 | 
  8 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
  9 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 10 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 11 |       To: sip:[tel]@[remote_ip]:[remote_port]
 12 |       Call-ID: [call_id]
 13 |       CSeq: [cseq] INVITE
 14 |       Contact: sip:[service]@[local_ip]:[local_port]
 15 |       Content-Type: application/sdp
 16 |       Content-Length: [len]
 17 | 
 18 |       v=0
 19 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 20 |       s=-
 21 |       c=IN IP[media_ip_type] [media_ip]
 22 |       t=0 0
 23 |       m=audio [media_port] RTP/AVP 8 0
 24 |       a=rtpmap:8 PCMA/8000
 25 |       a=rtpmap:0 PCMU/8000
 26 | 
 27 |     ]]>
 28 |   </send>
 29 | 
 30 |   <recv response="100" optional="true" response_txn="invite"/>
 31 | 
 32 |   <recv response="401" optional="true" auth="true" next="invite-with-auth" response_txn="invite">
 33 |     <action>
 34 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 35 |     </action>
 36 |   </recv>
 37 | 
 38 |   <recv response="407" optional="true" auth="true" next="invite-with-auth" response_txn="invite">
 39 |     <action>
 40 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 41 |     </action>
 42 |   </recv>
 43 | 
 44 |   <recv response="180" optional="true" response_txn="invite"/>
 45 | 
 46 |   <recv response="183" optional="true" response_txn="invite"/>
 47 | 
 48 |   <recv response="200" next="invite-ack" rrs="true" response_txn="invite"/>
 49 | 
 50 |   <label id="invite-with-auth"/>
 51 | 
 52 |   <!-- ACK is within transaction because of non-2xx. Reuse Via branch.      -->
 53 |   <!-- We don't use last_Via: because it would include rport stuff from     -->
 54 |   <!-- remote.                                                              -->
 55 |   <send ack_txn="invite">
 56 |     <![CDATA[
 57 | 
 58 |       ACK sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 59 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$1]
 60 |       [last_From:]
 61 |       [last_To:]
 62 |       [last_Call-ID:]
 63 |       CSeq: [cseq] ACK
 64 |       Contact: sip:[service]@[local_ip]:[local_port]
 65 |       Content-Length: 0
 66 | 
 67 |     ]]>
 68 |   </send>
 69 | 
 70 |   <send retrans="500" start_txn="invite">
 71 |     <![CDATA[
 72 | 
 73 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 74 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 75 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 76 |       To: sip:[tel]@[remote_ip]:[remote_port]
 77 |       [authentication]
 78 |       Call-ID: [call_id]
 79 |       CSeq: [cseq] INVITE
 80 |       Contact: sip:[service]@[local_ip]:[local_port]
 81 |       Content-Type: application/sdp
 82 |       Content-Length: [len]
 83 | 
 84 |       v=0
 85 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 86 |       s=-
 87 |       c=IN IP[media_ip_type] [media_ip]
 88 |       t=0 0
 89 |       m=audio [media_port] RTP/AVP 8 0
 90 |       a=rtpmap:8 PCMA/8000
 91 |       a=rtpmap:0 PCMU/8000
 92 | 
 93 |     ]]>
 94 |   </send>
 95 | 
 96 |   <recv response="100" response_txn="invite"/>
 97 | 
 98 |   <recv response="180" optional="true" response_txn="invite"/>
 99 | 
100 |   <recv response="183" optional="true" response_txn="invite"/>
101 | 
102 |   <recv response="200" rrs="true" response_txn="invite"/>
103 | 
104 |   <label id="invite-ack"/>
105 | 
106 |   <send ack_txn="invite">
107 |     <![CDATA[
108 | 
109 |       ACK [next_url] SIP/2.0
110 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
111 |       [routes]
112 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
113 |       To: sip:[tel]@[remote_ip]:[remote_port][peer_tag_param]
114 |       Call-ID: [call_id]
115 |       CSeq: [cseq] ACK
116 |       Contact: sip:[service]@[local_ip]:[local_port]
117 |       Content-Length: 0
118 | 
119 |     ]]>
120 |   </send>
121 | 
122 |   <!-- Send some audio to avoid rtptimeout timers kicking in. -->
123 |   <nop>
124 |     <action>
125 |       <exec rtp_stream="beep_1sec_50x160b.alaw,-1,8"/>
126 |     </action>
127 |   </nop>
128 | 
129 |   <recv request="BYE"/>
130 | 
131 |   <send>
132 |     <![CDATA[
133 | 
134 |       SIP/2.0 200 OK
135 |       [last_Via:]
136 |       [last_Record-Route:]
137 |       [last_From:]
138 |       [last_To:]
139 |       [last_Call-ID:]
140 |       [last_CSeq:]
141 |       Content-Length: 0
142 | 
143 |     ]]>
144 |   </send>
145 | 
146 |   <timewait milliseconds="2000"/><!-- retransmit 200's for a while -->
147 | 
148 | </scenario>
149 | 


--------------------------------------------------------------------------------
/out-in/oinv-i200-iinv-o200-ibye.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="oinv-i200-iinv-o200-ibye.xml -- wjd 2015">
  4 | 
  5 |   <send retrans="500" start_txn="invite">
  6 |     <![CDATA[
  7 | 
  8 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
  9 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 10 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 11 |       To: sip:[tel]@[remote_ip]:[remote_port]
 12 |       Call-ID: [call_id]
 13 |       CSeq: [cseq] INVITE
 14 |       Contact: sip:[service]@[local_ip]:[local_port]
 15 |       Content-Type: application/sdp
 16 |       Content-Length: [len]
 17 | 
 18 |       v=0
 19 |       o=user1 12345678 1234567890 IN IP[local_ip_type] [local_ip]
 20 |       s=-
 21 |       c=IN IP[media_ip_type] [media_ip]
 22 |       t=0 0
 23 |       m=audio [media_port] RTP/AVP 8 9
 24 |       a=rtpmap:8 PCMA/8000
 25 |       a=rtpmap:9 G722/8000
 26 | 
 27 |     ]]>
 28 |   </send>
 29 | 
 30 |   <recv response="100" optional="true" response_txn="invite"/>
 31 | 
 32 |   <recv response="401" optional="true" auth="true" next="invite-with-auth" response_txn="invite">
 33 |     <action>
 34 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 35 |     </action>
 36 |   </recv>
 37 | 
 38 |   <recv response="407" auth="true" next="invite-with-auth" response_txn="invite">
 39 |     <action>
 40 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 41 |     </action>
 42 |   </recv>
 43 | 
 44 |   <label id="invite-with-auth"/>
 45 | 
 46 |   <!-- ACK is within transaction because of non-2xx. Reuse Via branch.      -->
 47 |   <!-- We don't use last_Via: because it would include rport stuff from     -->
 48 |   <!-- remote.                                                              -->
 49 |   <send ack_txn="invite">
 50 |     <![CDATA[
 51 | 
 52 |       ACK sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 53 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$1]
 54 |       [last_From:]
 55 |       [last_To:]
 56 |       [last_Call-ID:]
 57 |       CSeq: [cseq] ACK
 58 |       Contact: sip:[service]@[local_ip]:[local_port]
 59 |       Content-Length: 0
 60 | 
 61 |     ]]>
 62 |   </send>
 63 | 
 64 |   <send retrans="500" start_txn="invite">
 65 |     <![CDATA[
 66 | 
 67 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 68 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 69 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 70 |       To: sip:[tel]@[remote_ip]:[remote_port]
 71 |       [authentication]
 72 |       Call-ID: [call_id]
 73 |       CSeq: [cseq] INVITE
 74 |       Contact: sip:[service]@[local_ip]:[local_port]
 75 |       Content-Type: application/sdp
 76 |       Content-Length: [len]
 77 | 
 78 |       v=0
 79 |       o=user1 12345678 1234567890 IN IP[local_ip_type] [local_ip]
 80 |       s=-
 81 |       c=IN IP[media_ip_type] [media_ip]
 82 |       t=0 0
 83 |       m=audio [media_port] RTP/AVP 8 9
 84 |       a=rtpmap:8 PCMA/8000
 85 |       a=rtpmap:9 G722/8000
 86 | 
 87 |     ]]>
 88 |   </send>
 89 | 
 90 |   <recv response="100" optional="true" response_txn="invite"/>
 91 | 
 92 |   <recv response="180" optional="true" response_txn="invite"/>
 93 | 
 94 |   <recv response="183" optional="true" response_txn="invite"/>
 95 | 
 96 |   <recv response="200" rrs="true" response_txn="invite"/>
 97 | 
 98 |   <send ack_txn="invite">
 99 |     <![CDATA[
100 | 
101 |       ACK [next_url] SIP/2.0
102 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
103 |       [routes]
104 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
105 |       To: sip:[tel]@[remote_ip]:[remote_port][peer_tag_param]
106 |       Call-ID: [call_id]
107 |       CSeq: [cseq] ACK
108 |       Contact: sip:[service]@[local_ip]:[local_port]
109 |       Content-Length: 0
110 | 
111 |     ]]>
112 |   </send>
113 | 
114 |   <nop>
115 |     <action>
116 |       <exec rtp_stream="beep_1sec_50x160b.alaw,1,8"/>
117 |     </action>
118 |   </nop>
119 | 
120 |   <recv request="INVITE" rrs="true"/>
121 | 
122 |   <!-- Update with only G722 codec. -->
123 |   <send retrans="500">
124 |     <![CDATA[
125 | 
126 |       SIP/2.0 200 OK
127 |       [last_Via:]
128 |       [last_Record-Route:]
129 |       [last_From:]
130 |       [last_To:];tag=[pid]SIPpTag01[call_number]
131 |       [last_Call-ID:]
132 |       [last_CSeq:]
133 |       Contact: sip:[service]@[local_ip]:[local_port]
134 |       Supported: replaces
135 |       Content-Type: application/sdp
136 |       Content-Length: [len]
137 | 
138 |       v=0
139 |       o=user1 12345678 1234567891 IN IP[local_ip_type] [local_ip]
140 |       s=-
141 |       c=IN IP[media_ip_type] [media_ip]
142 |       t=0 0
143 |       m=audio [media_port] RTP/AVP 9
144 |       a=rtpmap:9 G722/8000
145 | 
146 |     ]]>
147 |   </send>
148 | 
149 |   <recv request="ACK" rrs="true"/>
150 | 
151 |   <nop>
152 |     <action>
153 |       <exec rtp_stream="beep_1sec_50x160b.alaw,1,8"/><!-- pretend G722, send A-law anyway -->
154 |     </action>
155 |   </nop>
156 | 
157 |   <recv request="BYE"/>
158 | 
159 |   <send>
160 |     <![CDATA[
161 | 
162 |       SIP/2.0 200 OK
163 |       [last_Via:]
164 |       [last_Record-Route:]
165 |       [last_From:]
166 |       [last_To:]
167 |       [last_Call-ID:]
168 |       [last_CSeq:]
169 |       Content-Length: 0
170 | 
171 |     ]]>
172 |   </send>
173 | 
174 |   <timewait milliseconds="2000"/><!-- retransmit 200's for a while -->
175 | 
176 | </scenario>
177 | <!-- vim: set ts=8 sw=2 sts=2 et ai: -->
178 | 


--------------------------------------------------------------------------------
/out-in/oinv-i200-oinv-i200-oi3pcc-orefer-obye.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="iinv-o200-oinv-i200-oi3pcc-orefer-obye.xml -- wjd 2015">
  4 | 
  5 |   <!--
  6 | 
  7 |   * Send invite.
  8 |   * Initiate hold (a=sendonly).
  9 |   * Poke 3pcc to start setting up a new leg.
 10 |   * Get other dialog info from 3pcc.
 11 |   * Refer other end of this call to the other dialog.
 12 |   * Hang up this leg.
 13 | 
 14 | //Remote-Party-ID: [rpid]
 15 |   -->
 16 | 
 17 |   <send retrans="500" start_txn="invite">
 18 |     <![CDATA[
 19 | 
 20 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 21 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 22 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 23 |       To: sip:[tel]@[remote_ip]:[remote_port]
 24 |       Call-ID: [call_id]
 25 |       CSeq: [cseq] INVITE
 26 |       Contact: sip:[service]@[local_ip]:[local_port]
 27 |       Content-Type: application/sdp
 28 |       Content-Length: [len]
 29 | 
 30 |       v=0
 31 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 32 |       s=-
 33 |       c=IN IP[media_ip_type] [media_ip]
 34 |       t=0 0
 35 |       m=audio [media_port] RTP/AVP 8 0
 36 |       a=rtpmap:8 PCMA/8000
 37 |       a=rtpmap:0 PCMU/8000
 38 | 
 39 |     ]]>
 40 |   </send>
 41 | 
 42 |   <recv response="100" optional="true" response_txn="invite"/>
 43 | 
 44 |   <recv response="401" optional="true" auth="true" next="invite-with-auth" response_txn="invite">
 45 |     <action>
 46 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 47 |     </action>
 48 |   </recv>
 49 | 
 50 |   <recv response="407" auth="true" response_txn="invite">
 51 |     <action>
 52 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 53 |     </action>
 54 |   </recv>
 55 | 
 56 |   <label id="invite-with-auth"/>
 57 | 
 58 |   <!-- ACK is within transaction because of non-2xx. Reuse Via branch.      -->
 59 |   <!-- We don't use last_Via: because it would include rport stuff from     -->
 60 |   <!-- remote.                                                              -->
 61 |   <send ack_txn="invite">
 62 |     <![CDATA[
 63 | 
 64 |       ACK sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 65 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$1]
 66 |       [last_From:]
 67 |       [last_To:]
 68 |       [last_Call-ID:]
 69 |       CSeq: [cseq] ACK
 70 |       Contact: sip:[service]@[local_ip]:[local_port]
 71 |       Content-Length: 0
 72 | 
 73 |     ]]>
 74 |   </send>
 75 | 
 76 |   <send retrans="500" start_txn="invite">
 77 |     <![CDATA[
 78 | 
 79 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 80 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 81 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 82 |       To: sip:[tel]@[remote_ip]:[remote_port]
 83 |       [authentication]
 84 |       Call-ID: [call_id]
 85 |       CSeq: [cseq] INVITE
 86 |       Contact: sip:[service]@[local_ip]:[local_port]
 87 |       Content-Type: application/sdp
 88 |       Content-Length: [len]
 89 | 
 90 |       v=0
 91 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 92 |       s=-
 93 |       c=IN IP[media_ip_type] [media_ip]
 94 |       t=0 0
 95 |       m=audio [media_port] RTP/AVP 8 0
 96 |       a=rtpmap:8 PCMA/8000
 97 |       a=rtpmap:0 PCMU/8000
 98 | 
 99 |     ]]>
100 |   </send>
101 | 
102 |   <recv response="100" response_txn="invite"/>
103 | 
104 |   <recv response="180" optional="true" response_txn="invite"/>
105 | 
106 |   <recv response="183" optional="true" response_txn="invite"/>
107 | 
108 |   <recv response="200" rrs="true" response_txn="invite"/>
109 | 
110 |   <send ack_txn="invite">
111 |     <![CDATA[
112 | 
113 |       ACK [next_url] SIP/2.0
114 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
115 |       [routes]
116 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
117 |       To: sip:[tel]@[remote_ip]:[remote_port][peer_tag_param]
118 |       Call-ID: [call_id]
119 |       CSeq: [cseq] ACK
120 |       Contact: sip:[service]@[local_ip]:[local_port]
121 |       Content-Length: 0
122 | 
123 |     ]]>
124 |   </send>
125 | 
126 |   <!-- Send some audio -->
127 |   <nop>
128 |     <action>
129 |       <exec rtp_stream="beep_1sec_50x160b.alaw,1,8"/>
130 |     </action>
131 |   </nop>
132 |   <pause milliseconds="1000"/>
133 | 
134 |   <!-- Initiate hold (a=sendonly) -->
135 |   <send retrans="500" start_txn="invite">
136 |     <![CDATA[
137 | 
138 |       INVITE [next_url] SIP/2.0
139 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
140 |       [routes]
141 |       [last_From:]
142 |       [last_To:]
143 |       Call-ID: [call_id]
144 |       CSeq: [cseq] INVITE
145 |       Contact: sip:[service]@[local_ip]:[local_port]
146 |       Content-Type: application/sdp
147 |       Content-Length: [len]
148 | 
149 |       v=0
150 |       o=user1 53655765 2353687638 IN IP[local_ip_type] [local_ip]
151 |       s=-
152 |       c=IN IP[media_ip_type] 0.0.0.0
153 |       t=0 0
154 |       m=audio [media_port] RTP/AVP 8 0
155 |       a=rtpmap:8 PCMA/8000
156 |       a=rtpmap:0 PCMU/8000
157 |       a=sendonly
158 | 
159 |     ]]>
160 |   </send>
161 | 
162 |   <recv response="100" optional="true" response_txn="invite"/>
163 | 
164 |   <recv response="200" rrs="true" response_txn="invite"/>
165 | 
166 |   <send ack_txn="invite">
167 |     <![CDATA[
168 | 
169 |       ACK [next_url] SIP/2.0
170 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
171 |       [routes]
172 |       [last_From:]
173 |       [last_To:]
174 |       Call-ID: [call_id]
175 |       CSeq: [cseq] ACK
176 |       Contact: sip:[service]@[local_ip]:[local_port]
177 |       Content-Length: 0
178 | 
179 |     ]]>
180 |   </send>
181 | 
182 |   <!-- Peer is placed on hold. Poke 3pcc to initiate a second dialog.
183 |        (Call-ID must be in there. It's a 3pcc limitation/feature.) -->
184 |   <sendCmd>
185 |     <![CDATA[
186 | 
187 |       Call-ID: [call_id]
188 | 
189 |       Poke the peer. (And tell them our Call-ID at the same time.)
190 | 
191 |     ]]>
192 |   </sendCmd>
193 | 
194 |   <!-- Wait for 3pcc to come up with dialog info that we can steal. -->
195 |   <recvCmd>
196 |     <action>
197 |       <!-- Store next_url in $3 -->
198 |       <ereg regexp="(sip:[^@]+@[^>;]+)"
199 |             search_in="hdr"
200 |             header="Contact:"
201 |             check_it="true"
202 |             assign_to="3"/>
203 |       <!-- Store last_From-tag in $4 -->
204 |       <ereg regexp="tag=([^;]*)"
205 |             search_in="hdr"
206 |             header="From:"
207 |             check_it="true"
208 |             assign_to="9,4"/>
209 |       <!-- Store last_To-tag in $5 -->
210 |       <ereg regexp="tag=([^;]*)"
211 |             search_in="hdr"
212 |             header="To:"
213 |             check_it="true"
214 |             assign_to="9,5"/>
215 |     </action>
216 |   </recvCmd>
217 | 
218 |   <!-- This wait-time signifies that we called our peer and are now
219 |        moving our hand to the XFER button. -->
220 |   <pause milliseconds="1000"/>
221 | 
222 |   <!-- Refer the new dialog to the old (our) dialog.
223 |        Note how we "dedup///" the call_id this time: the peer
224 |        uses "dedup///[call_id]" as their call id.  -->
225 |   <send>
226 |     <![CDATA[
227 | 
228 |       REFER [next_url] SIP/2.0
229 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
230 |       [routes]
231 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
232 |       To: sip:[tel]@[remote_ip]:[remote_port][peer_tag_param]
233 |       Call-ID: [call_id]
234 |       CSeq: [cseq] REFER
235 |       Contact: sip:[service]@[local_ip]:[local_port]
236 |       Referred-By: sip:[service]@[local_ip]:[local_port]
237 |       Refer-To: <[$3]?Replaces=dedup///[call_id]%3Bto-tag%3D[$5]%3Bfrom-tag%3D[$4]>
238 |       Content-Length: 0
239 | 
240 |   ]]>
241 |   </send>
242 | 
243 |   <!-- Sometimes the 202 comes before the notify, sometimes after.
244 |        We'll allow an optional NOTIFY first, and wait for the retransmit to
245 |        answer it. -->
246 |   <recv request="NOTIFY" optional="true"/>
247 |   <recv response="202"/>
248 | 
249 |   <!-- This should contain a 200 success message, or 180 ringing,
250 |        depending on the other scenarios. -->
251 |   <recv request="NOTIFY"/>
252 | 
253 |   <send>
254 |     <![CDATA[
255 | 
256 |       SIP/2.0 200 OK
257 |       [last_Via:]
258 |       [last_From:]
259 |       [last_To:]
260 |       [last_Call-ID:]
261 |       [last_CSeq:]
262 |       [last_Event:]
263 |       Contact: sip:[service]@[local_ip]:[local_port]
264 |       Content-Length: 0
265 | 
266 |     ]]>
267 |   </send>
268 | 
269 |   <!-- Hang up this old leg. -->
270 |   <send start_txn="bye">
271 |     <![CDATA[
272 | 
273 |       BYE [next_url] SIP/2.0
274 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
275 |       [routes]
276 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
277 |       To: sip:[tel]@[remote_ip]:[remote_port][peer_tag_param]
278 |       [last_Call-ID:]
279 |       CSeq: [cseq] BYE
280 |       Contact: sip:[service]@[local_ip]:[local_port]
281 |       Content-Length: 0
282 | 
283 |     ]]>
284 |   </send>
285 | 
286 |   <recv response="200" response_txn="bye"/>
287 | 
288 | </scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->
289 | 


--------------------------------------------------------------------------------
/out-in/oinv-i486.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="UTF-8" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | <scenario name="oinv-i486.xml -- wjd 2011-2012">
  4 | 
  5 |   <label id="invite"/>
  6 | 
  7 |   <send retrans="500" start_txn="invite">
  8 |     <![CDATA[
  9 | 
 10 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 11 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 12 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 13 |       To: sip:[tel]@[remote_ip]:[remote_port]
 14 |       Contact: sip:[service]@[local_ip]:[local_port]
 15 |       Call-ID: [call_id]
 16 |       CSeq: [cseq] INVITE
 17 |       Max-Forwards: 70
 18 |       Content-Type: application/sdp
 19 |       Content-Length: [len]
 20 | 
 21 |       v=0
 22 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 23 |       s=-
 24 |       c=IN IP[media_ip_type] [media_ip]
 25 |       t=0 0
 26 |       m=audio [media_port] RTP/AVP 8 0
 27 |       a=rtpmap:8 PCMA/8000
 28 |       a=rtpmap:0 PCMU/8000
 29 | 
 30 |     ]]>
 31 |   </send>
 32 | 
 33 |   <recv response="100" optional="true" response_txn="invite"/>
 34 | 
 35 |   <recv response="401" optional="true" next="invite-with-auth" auth="true" response_txn="invite">
 36 |     <action>
 37 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 38 |       <ereg regexp=";.*" search_in="hdr" header="To" check_it="false" assign_to="2"/>
 39 |     </action>
 40 |   </recv>
 41 | 
 42 |   <recv response="407" auth="true" response_txn="invite">
 43 |     <action>
 44 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
 45 |       <ereg regexp=";.*" search_in="hdr" header="To" check_it="false" assign_to="2"/>
 46 |     </action>
 47 |   </recv>
 48 | 
 49 |   <label id="invite-with-auth"/>
 50 | 
 51 |   <send ack_txn="invite">
 52 |     <![CDATA[
 53 | 
 54 |       ACK sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 55 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$1]
 56 |       [routes]
 57 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 58 |       To: sip:[tel]@[remote_ip]:[remote_port][$2]
 59 |       Contact: sip:[service]@[local_ip]:[local_port]
 60 |       Call-ID: [call_id]
 61 |       CSeq: [cseq] ACK
 62 |       Max-Forwards: 70
 63 |       Content-Length: 0
 64 | 
 65 |     ]]>
 66 |   </send>
 67 | 
 68 |   <send retrans="500" start_txn="invite">
 69 |     <![CDATA[
 70 | 
 71 |       INVITE sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
 72 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 73 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
 74 |       To: sip:[tel]@[remote_ip]:[remote_port]
 75 |       Contact: sip:[service]@[local_ip]:[local_port]
 76 |       [authentication]
 77 |       Call-ID: [call_id]
 78 |       CSeq: [cseq] INVITE
 79 |       Max-Forwards: 70
 80 |       Content-Type: application/sdp
 81 |       Content-Length: [len]
 82 | 
 83 |       v=0
 84 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 85 |       s=-
 86 |       c=IN IP[media_ip_type] [media_ip]
 87 |       t=0 0
 88 |       m=audio [media_port] RTP/AVP 8 0
 89 |       a=rtpmap:8 PCMA/8000
 90 |       a=rtpmap:0 PCMU/8000
 91 | 
 92 |     ]]>
 93 |   </send>
 94 | 
 95 |   <recv response="100" response_txn="invite"/>
 96 | 
 97 |   <!-- Call is being forwarded -->
 98 |   <recv response="181" optional="true" response_txn="invite"/>
 99 | 
100 |   <recv response="180" optional="true" response_txn="invite"/>
101 | 
102 |   <recv response="183" optional="true" response_txn="invite"/>
103 | 
104 |   <recv response="486" auth="true" response_txn="invite">
105 |     <action>
106 |       <ereg regexp=";branch=[^;]*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
107 |       <ereg regexp=";.*" search_in="hdr" header="To" check_it="false" assign_to="2"/>
108 |     </action>
109 |   </recv>
110 | 
111 |   <send ack_txn="invite">
112 |     <![CDATA[
113 | 
114 |       ACK sip:[tel]@[remote_ip]:[remote_port] SIP/2.0
115 |       Via: SIP/2.0/[transport] [local_ip]:[local_port][$1]
116 |       [routes]
117 |       From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
118 |       To: sip:[tel]@[remote_ip]:[remote_port][$2]
119 |       Contact: sip:[service]@[local_ip]:[local_port]
120 |       Call-ID: [call_id]
121 |       CSeq: [cseq] ACK
122 |       Max-Forwards: 70
123 |       Content-Length: 0
124 | 
125 |     ]]>
126 |   </send>
127 | 
128 | </scenario>
129 | 


--------------------------------------------------------------------------------
/out-in/oreg-i200.xml:
--------------------------------------------------------------------------------
 1 | <?xml version="1.0" encoding="UTF-8"?>
 2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
 3 | <scenario name="oreg-i200.xml -- wjd 2011">
 4 |   <label id="register"/>
 5 | 
 6 |   <send retrans="500">
 7 |     <![CDATA[
 8 | 
 9 |       REGISTER sip:[remote_ip]:[remote_port] SIP/2.0
10 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
11 |       From: <sip:[service]@[remote_ip]:[remote_port]>;tag=[pid]SIPpTag00[call_number]
12 |       To: <sip:[service]@[remote_ip]:[remote_port]>
13 |       Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
14 |       Expires: 15
15 |       Call-ID: [call_id]
16 |       CSeq: 1 REGISTER
17 |       Content-Length: 0
18 | 
19 |     ]]>
20 |   </send>
21 | 
22 |   <recv response="200" optional="true" next="register-done"/>
23 | 
24 |   <recv response="401" optional="true" next="register-with-auth" auth="true"/>
25 | 
26 |   <recv response="407" auth="true"/>
27 | 
28 |   <label id="register-with-auth"/>
29 | 
30 |   <send retrans="500">
31 |     <![CDATA[
32 | 
33 |       REGISTER sip:[remote_ip]:[remote_port] SIP/2.0
34 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
35 |       From: <sip:[service]@[remote_ip]:[remote_port]>;tag=[pid]SIPpTag00[call_number]
36 |       To: <sip:[service]@[remote_ip]:[remote_port]>
37 |       Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
38 |       Expires: 15
39 |       [authentication]
40 |       Call-ID: [call_id]
41 |       CSeq: 2 REGISTER
42 |       Content-Length: 0
43 | 
44 |     ]]>
45 |   </send>
46 | 
47 |   <recv response="200"/>
48 | 
49 |   <label id="register-done"/>
50 | </scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->
51 | 


--------------------------------------------------------------------------------