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 | <!-- --> |
| 20 | |
| 21 | <!-- Note: |
| 22 | For this test to work, PJSUA-LIB needs to add video line, with |
| 23 | this patch: |
| 24 | |
| 25 | pjsua_media.c:1253, after call to pjmedia_endpt_create_sdp(): |
| 26 | |
| 27 | if (1) { |
| 28 | pjmedia_sdp_media *m = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_media); |
| 29 | m->desc.media = pj_str("video"); |
| 30 | m->desc.port = 3000; |
| 31 | m->desc.transport = pj_str("RTP/AVP"); |
| 32 | m->desc.fmt_count = 1; |
| 33 | m->desc.fmt[0] = pj_str("0"); |
| 34 | sdp->media[sdp->media_count++] = m; |
| 35 | } |
| 36 | |
| 37 | --> |
| 38 | |
| 39 | |
| 40 | <scenario name="UAC with bad ACK"> |
| 41 | <!-- UAC with bad ACK causes assertion with pjsip 1.4 --> |
| 42 | <send retrans="500"> |
| 43 | <![CDATA[ |
| 44 | |
| 45 | INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 |
| 46 | Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] |
| 47 | From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] |
| 48 | To: sut <sip:[service]@[remote_ip]:[remote_port]> |
| 49 | Call-ID: [call_id] |
| 50 | CSeq: 1 INVITE |
| 51 | Contact: sip:sipp@[local_ip]:[local_port] |
| 52 | Max-Forwards: 70 |
| 53 | Subject: Performance Test |
| 54 | Content-Type: application/sdp |
| 55 | Content-Length: [len] |
| 56 | |
| 57 | v=0 |
| 58 | o=Tester 234 123 IN IP4 89.208.145.194 |
| 59 | s=Tester |
| 60 | c=IN IP4 89.208.145.194 |
| 61 | t=0 0 |
| 62 | m=audio 17424 RTP/AVP 111 0 18 101 |
| 63 | a=rtpmap:111 SPEEX/16000 |
| 64 | a=rtpmap:0 PCMU/8000 |
| 65 | a=rtpmap:18 G729/8000 |
| 66 | a=rtpmap:101 telephone-event/8000 |
| 67 | a=sendrecv |
| 68 | a=rtcp:17425 |
| 69 | m=video 11128 RTP/AVP 34 103 104 |
| 70 | a=rtpmap:34 H263/90000 |
| 71 | a=rtpmap:103 H263-1998/90000 |
| 72 | a=rtpmap:104 H264/90000 |
| 73 | a=sendrecv |
| 74 | a=rtcp:11129 |
| 75 | |
| 76 | ]]> |
| 77 | </send> |
| 78 | |
| 79 | <recv response="100" |
| 80 | optional="true"> |
| 81 | </recv> |
| 82 | |
| 83 | <recv response="180" optional="true"> |
| 84 | </recv> |
| 85 | |
| 86 | <!-- By adding rrs="true" (Record Route Sets), the route sets --> |
| 87 | <!-- are saved and used for following messages sent. Useful to test --> |
| 88 | <!-- against stateful SIP proxies/B2BUAs. --> |
| 89 | <recv response="200" rtd="true"> |
| 90 | </recv> |
| 91 | |
| 92 | <!-- Packet lost can be simulated in any send/recv message by --> |
| 93 | <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --> |
| 94 | <send> |
| 95 | <![CDATA[ |
| 96 | |
| 97 | ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 |
| 98 | Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] |
| 99 | From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] |
| 100 | To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] |
| 101 | Call-ID: [call_id] |
| 102 | CSeq: 1 ACK |
| 103 | Contact: sip:sipp@[local_ip]:[local_port] |
| 104 | Max-Forwards: 70 |
| 105 | Subject: Performance Test |
| 106 | Content-Length: 0 |
| 107 | |
| 108 | ]]> |
| 109 | </send> |
| 110 | |
| 111 | <!-- This delay can be customized by the -d command-line option --> |
| 112 | <!-- or by adding a 'milliseconds = "value"' option here. --> |
| 113 | <pause milliseconds="2000"/> |
| 114 | |
| 115 | <send retrans="500"> |
| 116 | <![CDATA[ |
| 117 | |
| 118 | INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 |
| 119 | Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] |
| 120 | From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] |
| 121 | To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] |
| 122 | Call-ID: [call_id] |
| 123 | CSeq: 2 INVITE |
| 124 | Contact: sip:sipp@[local_ip]:[local_port] |
| 125 | Max-Forwards: 70 |
| 126 | Subject: Performance Test |
| 127 | Content-Type: application/sdp |
| 128 | Content-Length: [len] |
| 129 | |
| 130 | v=0 |
| 131 | o=Tester 234 124 IN IP4 89.208.145.194 |
| 132 | s=Tester |
| 133 | c=IN IP4 89.208.145.194 |
| 134 | t=0 0 |
| 135 | m=audio 17424 RTP/AVP 111 0 18 101 |
| 136 | a=rtpmap:111 SPEEX/16000 |
| 137 | a=rtpmap:0 PCMU/8000 |
| 138 | a=rtpmap:18 G729/8000 |
| 139 | a=rtpmap:101 telephone-event/8000 |
| 140 | a=sendrecv |
| 141 | a=rtcp:17425 |
| 142 | m=video 0 RTP/AVP 34 103 104 |
| 143 | a=sendrecv |
| 144 | |
| 145 | |
| 146 | ]]> |
| 147 | </send> |
| 148 | |
| 149 | <!-- By adding rrs="true" (Record Route Sets), the route sets --> |
| 150 | <!-- are saved and used for following messages sent. Useful to test --> |
| 151 | <!-- against stateful SIP proxies/B2BUAs. --> |
| 152 | <recv response="200" rtd="true"> |
| 153 | </recv> |
| 154 | |
| 155 | <!-- Packet lost can be simulated in any send/recv message by --> |
| 156 | <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --> |
| 157 | <send> |
| 158 | <![CDATA[ |
| 159 | |
| 160 | ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 |
| 161 | Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] |
| 162 | From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] |
| 163 | To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] |
| 164 | Call-ID: [call_id] |
| 165 | CSeq: 2 ACK |
| 166 | Contact: sip:sipp@[local_ip]:[local_port] |
| 167 | Max-Forwards: 70 |
| 168 | Subject: Performance Test |
| 169 | Content-Length: 0 |
| 170 | |
| 171 | ]]> |
| 172 | </send> |
| 173 | |
| 174 | |
| 175 | <pause milliseconds="2000"/> |
| 176 | |
| 177 | |
| 178 | <!-- The 'crlf' option inserts a blank line in the statistics report. --> |
| 179 | <send retrans="500"> |
| 180 | <![CDATA[ |
| 181 | |
| 182 | BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 |
| 183 | Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] |
| 184 | From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] |
| 185 | To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] |
| 186 | Call-ID: [call_id] |
| 187 | CSeq: 3 BYE |
| 188 | Contact: sip:sipp@[local_ip]:[local_port] |
| 189 | Max-Forwards: 70 |
| 190 | Subject: Performance Test |
| 191 | Content-Length: 0 |
| 192 | |
| 193 | ]]> |
| 194 | </send> |
| 195 | |
| 196 | <recv response="200" crlf="true"> |
| 197 | </recv> |
| 198 | |
| 199 | |
| 200 | <!-- definition of the response time repartition table (unit is ms) --> |
| 201 | <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/> |
| 202 | |
| 203 | <!-- definition of the call length repartition table (unit is ms) --> |
| 204 | <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> |
| 205 | |
| 206 | </scenario> |
| 207 | |