diff options
Diffstat (limited to 'tests/pjsua/scripts-sipp/uas-timer-reinvite.xml')
| -rw-r--r-- | tests/pjsua/scripts-sipp/uas-timer-reinvite.xml | 108 |
1 files changed, 108 insertions, 0 deletions
diff --git a/tests/pjsua/scripts-sipp/uas-timer-reinvite.xml b/tests/pjsua/scripts-sipp/uas-timer-reinvite.xml new file mode 100644 index 0000000..fe5169b --- /dev/null +++ b/tests/pjsua/scripts-sipp/uas-timer-reinvite.xml @@ -0,0 +1,108 @@ +<?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> + |