[asterisk-users] RTCP not being sent when on hold
adrianvoip at gmail.com
Wed Apr 9 16:01:43 CDT 2008
The RTP codec 126 is a bogus RTP packet sent by Bria to maintain the NAT
I've identified the issue as this:
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
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?
On Wed, Apr 9, 2008 at 7:15 AM, Steve Langstaff <steve.langstaff at citel.com>
> It would be interesting to see a wireshark trace of the SIP and RTP
> traffic during call setup and hold, to see:
> a) what codec 126 has been negotiated as and
> b) who is sourcing the unknown RTP datagram.
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Adrian A
> *Sent:* 09 April 2008 00:55
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] RTCP not being sent when on hold
> When I receive a call to my CounterPath Bria from Asterisk 184.108.40.206 and I
> place the call on hold, the call is dropped after 30 seconds.
> 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:
> NOTICE rtp.c: Unknown RTP codec 126 received from '0.0.0.0'
> In sip.conf I have rtpkeepalive=15 but that does not seem to help.
> Does anyone know what I can do to fix this, other than increase the
> timeout on Bria?
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