Tristan Matthews | 0a329cc | 2013-07-17 13:20:14 -0400 | [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="UAS answer 200/INVITE without SDP (#1045)"> |
| 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 | </recv> |
| 29 | |
| 30 | <send retrans="500"> |
| 31 | <![CDATA[ |
| 32 | |
| 33 | SIP/2.0 200 OK |
| 34 | [last_Via:] |
| 35 | [last_From:] |
| 36 | [last_To:];tag=[call_number] |
| 37 | [last_Call-ID:] |
| 38 | [last_CSeq:] |
| 39 | Contact: sip:sipp@[local_ip]:[local_port] |
| 40 | Content-Type: application/sdp |
| 41 | Content-Length: [len] |
| 42 | |
| 43 | ]]> |
| 44 | </send> |
| 45 | |
| 46 | <recv request="ACK" crlf="true"> |
| 47 | </recv> |
| 48 | |
| 49 | |
| 50 | <recv request="BYE" crlf="true"> |
| 51 | </recv> |
| 52 | |
| 53 | <send> |
| 54 | <![CDATA[ |
| 55 | |
| 56 | SIP/2.0 200 OK |
| 57 | [last_Via:] |
| 58 | [last_From:] |
| 59 | [last_To:] |
| 60 | [last_Call-ID:] |
| 61 | [last_CSeq:] |
| 62 | Contact: sip:sipp@[local_ip]:[local_port] |
| 63 | Content-Length: [len] |
| 64 | |
| 65 | ]]> |
| 66 | </send> |
| 67 | |
| 68 | <!-- Keep the call open for a while in case the 200 is lost to be --> |
| 69 | <!-- able to retransmit it if we receive the BYE again. --> |
| 70 | <pause milliseconds="4000"/> |
| 71 | |
| 72 | |
| 73 | <!-- definition of the response time repartition table (unit is ms) --> |
| 74 | <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/> |
| 75 | |
| 76 | <!-- definition of the call length repartition table (unit is ms) --> |
| 77 | <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> |
| 78 | |
| 79 | </scenario> |
| 80 | |