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gordon+asterisk at dro... Guest
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Posted: Wed Apr 09, 2008 4:15 am Post subject: [asterisk-users] RTCP not being sent when on hold |
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On Tue, 8 Apr 2008, Adrian A wrote:
Quote: | 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?
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Are you also recording the call?
I had to put this:
[options]
transmit_silence_during_record = yes
into asterisk.conf to stop hangups after 30 seconds ...
Gordon |
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drew at oanda.com Guest
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Posted: Wed Apr 09, 2008 8:16 am Post subject: [asterisk-users] RTCP not being sent when on hold |
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Adrian A wrote:
Quote: | Hello,
When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1
<http://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
<http://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
| Is it not up to the phone to send the keep-alive packets?
Sounds like Asterisk does not understand the keep-alive packets coming
from the phone. Try setting "rtptimeout=300" in sip.conf to test this.
It should now hangup after 5 minutes.
regards,
Drew
regards,
Drew
--
Drew Gibson
Systems Administrator
OANDA Corporation
www.oanda.com |
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steve.langstaff at cit... Guest
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Posted: Wed Apr 09, 2008 9:15 am Post subject: [asterisk-users] RTCP not being sent when on hold |
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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
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adrianvoip at gmail.com Guest
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Posted: Wed Apr 09, 2008 4:01 pm Post subject: [asterisk-users] RTCP not being sent when on hold |
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The RTP codec 126 is a bogus RTP packet sent by Bria to maintain the NAT
binding.
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
time.
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>
wrote:
Quote: | 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
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