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Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.

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Review: https://reviewboard.asterisk.org/r/1891/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2012-07-01 17:28:57 +00:00
parent 628425ba6f
commit 37256ea45d
1947 changed files with 1120896 additions and 37 deletions
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126
res/pjproject/tests/pjsua/scripts-sipp/uas-timer-update.xml Normal file
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<?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 incidates support for UPDATE. -->
<!-- In this case, UAC will first use empty UPDATE, which we -->
<!-- will reply with 400. UAC MUST retry sending UPDATE 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]>
Allow: UPDATE, INVITE
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="UPDATE" crlf="true">
</recv>
<send>
<![CDATA[
SIP/2.0 400 Want SDP Body
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Allow: INVITE
Require: timer
Session-Expires: 90;refresher=uac
Content-Length: 0
]]>
</send>
<recv request="UPDATE" crlf="true">
</recv>
<send>
<![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]>
Allow: INVITE
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>
<!-- 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>