The RTP codec 126 is a bogus RTP packet sent by Bria to maintain the NAT binding.<br><br>I've identified the issue as this:<br><br>Bria has an inactivity timer that is based on RTCP. Basically, if during the call there is RTCP, Bria uses it to make sure the call is still alive. Asterisk does send RTCP when call is active, but it stops when call is put on hold by Bria. The default timeout for Bria is 30 seconds, thus it disconnects the call because it has not received any RTP or RTCP during this time.<br>
<br>I am not sure at this point which is correct implementation. Should the client not rely on RTP/RTCP when it's on hold or should Asterisk send some sort of keep alive RTP/RTCP when it knows one of the clients is on hold?<br>
<br><br><div class="gmail_quote">On Wed, Apr 9, 2008 at 7:15 AM, Steve Langstaff <<a href="mailto:steve.langstaff@citel.com">steve.langstaff@citel.com</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2">It would be interesting to see a wireshark trace of
the SIP and RTP traffic during call setup and hold, to
see:</font></span></div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2">a) what codec 126 has been negotiated as
and</font></span></div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2">b) who is sourcing the unknown RTP
datagram.</font></span></div><br>
<blockquote style="border-left: 2px solid rgb(0, 0, 255); padding-left: 5px; margin-left: 5px; margin-right: 0px;">
<div dir="ltr" align="left" lang="en-us">
<hr>
<font face="Tahoma" size="2"><b>From:</b> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Adrian
A<br><b>Sent:</b> 09 April 2008 00:55<br><b>To:</b> Asterisk Users Mailing
List - Non-Commercial Discussion<br><b>Subject:</b> [asterisk-users] RTCP not
being sent when on hold<br></font><br></div><div><div></div><div class="Wj3C7c">
<div></div>Hello,<br><br>When I receive a call to my CounterPath Bria from
Asterisk <a href="http://1.4.18.1" target="_blank">1.4.18.1</a> and I place the call on hold,
the call is dropped after 30 seconds.<br>It looks like there is no RTCP/RTP
sent to the client from Asterisk while on hold (music on hold playing to
caller) thus client disconnects the call. During this time, I get the
following messages in the CLI:<br><br>NOTICE[24194] rtp.c: Unknown RTP codec
126 received from '<a href="http://0.0.0.0" target="_blank">0.0.0.0</a>'<br><br>In sip.conf I
have rtpkeepalive=15 but that does not seem to help.<br><br>Does anyone know
what I can do to fix this, other than increase the timeout on
Bria?<br><br>Thanks,<br>Adrian<br></div></div></blockquote></div>
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