| <?xml version="1.0" encoding="ISO-8859-1" ?> |
| <!DOCTYPE scenario SYSTEM "sipp.dtd"> |
| |
| <!-- This program is free software; you can redistribute it and/or --> |
| <!-- modify it under the terms of the GNU General Public License as --> |
| <!-- published by the Free Software Foundation; either version 2 of the --> |
| <!-- License, or (at your option) any later version. --> |
| <!-- --> |
| <!-- This program is distributed in the hope that it will be useful, --> |
| <!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --> |
| <!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --> |
| <!-- GNU General Public License for more details. --> |
| <!-- --> |
| <!-- You should have received a copy of the GNU General Public License --> |
| <!-- along with this program; if not, write to the --> |
| <!-- Free Software Foundation, Inc., --> |
| <!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --> |
| |
| |
| <!-- --> |
| <!-- Session timer where UAS doesn't indicate support for UPDATE. --> |
| <!-- In this case, UAC MUST use re-INVITE with SDP. --> |
| |
| <scenario name="Basic UAS responder"> |
| <recv request="INVITE" crlf="true"> |
| </recv> |
| |
| <send retrans="500"> |
| <![CDATA[ |
| |
| SIP/2.0 200 OK |
| [last_Via:] |
| [last_From:] |
| [last_To:];tag=[call_number] |
| [last_Call-ID:] |
| [last_CSeq:] |
| Contact: <sip:[local_ip]:[local_port];transport=[transport]> |
| Require: timer |
| Session-Expires: 90;refresher=uac |
| Content-Type: application/sdp |
| Content-Length: [len] |
| |
| v=0 |
| o=Some-UserAgent 68 210 IN IP4 [local_ip] |
| s=SIP Call |
| c=IN IP4 [local_ip] |
| t=0 0 |
| m=audio 17294 RTP/AVP 0 101 |
| c=IN IP4 [local_ip] |
| a=rtpmap:101 telephone-event/8000 |
| a=fmtp:101 0-16 |
| ]]> |
| </send> |
| |
| <recv request="ACK" |
| optional="true" |
| rtd="true" |
| crlf="true"> |
| </recv> |
| |
| <recv request="INVITE" crlf="true"> |
| </recv> |
| |
| <send retrans="500"> |
| <![CDATA[ |
| |
| SIP/2.0 200 OK |
| [last_Via:] |
| [last_From:] |
| [last_To:];tag=[call_number] |
| [last_Call-ID:] |
| [last_CSeq:] |
| Contact: <sip:[local_ip]:[local_port];transport=[transport]> |
| Require: timer |
| Session-Expires: 90;refresher=uac |
| Content-Type: application/sdp |
| Content-Length: [len] |
| |
| v=0 |
| o=Some-UserAgent 68 210 IN IP4 [local_ip] |
| s=SIP Call |
| c=IN IP4 [local_ip] |
| t=0 0 |
| m=audio 17294 RTP/AVP 0 101 |
| c=IN IP4 [local_ip] |
| a=rtpmap:101 telephone-event/8000 |
| a=fmtp:101 0-16 |
| ]]> |
| </send> |
| |
| <recv request="ACK" |
| rtd="true" |
| crlf="true"> |
| </recv> |
| |
| |
| <!-- Keep the call open for a while in case the 200 is lost to be --> |
| <!-- able to retransmit it if we receive the BYE again. --> |
| <pause milliseconds="4000"/> |
| |
| <!-- definition of the response time repartition table (unit is ms) --> |
| <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/> |
| |
| <!-- definition of the call length repartition table (unit is ms) --> |
| <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> |
| |
| </scenario> |
| |