Alexandre Lision | 8af73cb | 2013-12-10 14:11:20 -0500 | [diff] [blame] | 1 | <?xml version="1.0" encoding="ISO-8859-1" ?> |
| 2 | <!DOCTYPE scenario SYSTEM "sipp.dtd"> |
| 3 | |
| 4 | <!-- This program is free software; you can redistribute it and/or --> |
| 5 | <!-- modify it under the terms of the GNU General Public License as --> |
| 6 | <!-- published by the Free Software Foundation; either version 2 of the --> |
| 7 | <!-- License, or (at your option) any later version. --> |
| 8 | <!-- --> |
| 9 | <!-- This program is distributed in the hope that it will be useful, --> |
| 10 | <!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --> |
| 11 | <!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --> |
| 12 | <!-- GNU General Public License for more details. --> |
| 13 | <!-- --> |
| 14 | <!-- You should have received a copy of the GNU General Public License --> |
| 15 | <!-- along with this program; if not, write to the --> |
| 16 | <!-- Free Software Foundation, Inc., --> |
| 17 | <!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --> |
| 18 | <!-- --> |
| 19 | <!-- Sipp default 'uas' scenario. --> |
| 20 | <!-- --> |
| 21 | |
| 22 | <scenario name="Sending OK and re-INVITE with less media (#16xx)"> |
| 23 | <!-- By adding rrs="true" (Record Route Sets), the route sets --> |
| 24 | <!-- are saved and used for following messages sent. Useful to test --> |
| 25 | <!-- against stateful SIP proxies/B2BUAs. --> |
| 26 | |
| 27 | <recv request="INVITE" crlf="true"> |
| 28 | <action> |
| 29 | <ereg regexp=".*" search_in="hdr" header="From" assign_to="3"/> |
| 30 | <ereg regexp="sip:(.*)>" search_in="hdr" header="Contact" assign_to="4,5"/> |
| 31 | <assign assign_to="4" variable="5" /> |
| 32 | </action> |
| 33 | </recv> |
| 34 | |
| 35 | <send retrans="500"> |
| 36 | <![CDATA[ |
| 37 | |
| 38 | SIP/2.0 200 OK |
| 39 | [last_Via:] |
| 40 | [last_From:] |
| 41 | [last_To:] |
| 42 | [last_Call-ID:] |
| 43 | [last_CSeq:] |
| 44 | Contact: sip:sipp@[local_ip]:[local_port] |
| 45 | Content-Type: application/sdp |
| 46 | Content-Length: [len] |
| 47 | |
| 48 | v=0 |
| 49 | o=- 3441953879 3441953879 IN IP4 192.168.0.15 |
| 50 | s=pjmedia |
| 51 | c=IN IP4 192.168.0.15 |
| 52 | t=0 0 |
| 53 | m=audio 4000 RTP/AVP 0 96 |
| 54 | a=rtpmap:0 PCMU/8000 |
| 55 | a=rtpmap:96 telephone-event/8000 |
| 56 | a=sendrecv |
| 57 | |
| 58 | ]]> |
| 59 | </send> |
| 60 | |
| 61 | <recv request="ACK" crlf="true"> |
| 62 | </recv> |
| 63 | |
| 64 | <pause milliseconds="2000"/> |
| 65 | |
| 66 | <send retrans="500"> |
| 67 | <![CDATA[ |
| 68 | |
| 69 | INVITE sip:[$5] SIP/2.0 |
| 70 | Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-same-branch |
| 71 | From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] |
| 72 | To[$3] |
| 73 | Call-ID: [call_id] |
| 74 | Cseq: 2 INVITE |
| 75 | Max-Forwards: 70 |
| 76 | Content-Type: application/sdp |
| 77 | Content-Length: [len] |
| 78 | |
| 79 | v=0 |
| 80 | o=- 3441953879 3441953879 IN IP4 192.168.0.15 |
| 81 | s=pjmedia |
| 82 | c=IN IP4 192.168.0.15 |
| 83 | t=0 0 |
| 84 | m=audio 4000 RTP/AVP 0 96 |
| 85 | a=rtpmap:0 PCMU/8000 |
| 86 | a=rtpmap:96 telephone-event/8000 |
| 87 | a=sendonly |
| 88 | |
| 89 | ]]> |
| 90 | </send> |
| 91 | |
| 92 | <recv response="200" rtd="true"> |
| 93 | </recv> |
| 94 | |
| 95 | <send> |
| 96 | <![CDATA[ |
| 97 | |
| 98 | ACK sip:[$5] SIP/2.0 |
| 99 | Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-same-branch |
| 100 | From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] |
| 101 | To[$3] |
| 102 | Call-ID: [call_id] |
| 103 | Cseq: 1 ACK |
| 104 | Contact: sip:sipp@[local_ip]:[local_port] |
| 105 | Max-Forwards: 70 |
| 106 | Content-Length: 0 |
| 107 | |
| 108 | ]]> |
| 109 | </send> |
| 110 | |
| 111 | <recv request="INVITE" crlf="true"> |
| 112 | </recv> |
| 113 | |
| 114 | <send retrans="500"> |
| 115 | <![CDATA[ |
| 116 | |
| 117 | SIP/2.0 200 OK |
| 118 | [last_Via:] |
| 119 | [last_From:] |
| 120 | [last_To:] |
| 121 | [last_Call-ID:] |
| 122 | [last_CSeq:] |
| 123 | Contact: sip:sipp@[local_ip]:[local_port] |
| 124 | Content-Type: application/sdp |
| 125 | Content-Length: [len] |
| 126 | |
| 127 | v=0 |
| 128 | o=- 3441953879 3441953879 IN IP4 192.168.0.15 |
| 129 | s=pjmedia |
| 130 | c=IN IP4 192.168.0.15 |
| 131 | t=0 0 |
| 132 | m=audio 4000 RTP/AVP 0 96 |
| 133 | a=rtpmap:0 PCMU/8000 |
| 134 | a=rtpmap:96 telephone-event/8000 |
| 135 | a=inactive |
| 136 | |
| 137 | ]]> |
| 138 | </send> |
| 139 | |
| 140 | <recv request="ACK" crlf="true"> |
| 141 | </recv> |
| 142 | |
| 143 | <!-- Keep the call open for a while in case the 200 is lost to be --> |
| 144 | <!-- able to retransmit it if we receive the BYE again. --> |
| 145 | <pause milliseconds="4000"/> |
| 146 | |
| 147 | |
| 148 | <!-- definition of the response time repartition table (unit is ms) --> |
| 149 | <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/> |
| 150 | |
| 151 | <!-- definition of the call length repartition table (unit is ms) --> |
| 152 | <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> |
| 153 | |
| 154 | </scenario> |
| 155 | |