[asterisk-users] RTCP not being sent when on hold

[Steve Langstaff]

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
	
	
	Hello,
	
	When I receive a call to my CounterPath Bria from Asterisk
1.4.18.1 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[24194] 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?
	
	Thanks,
	Adrian
	

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080409/7a03a042/attachment.htm