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Summary:ASTERISK-12749: Loss of incoming audio during a phone call through a SIP trunk
Reporter:Alfredo Zambrano (azambrano)Labels:
Date Opened:2008-09-17 17:07:08Date Closed:2011-06-07 14:02:45
Priority:MajorRegression?No
Status:Closed/CompleteComponents:General
Versions:Frequency of
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Description:When I dial to a phone number through a sip trunk, there are some times where I loos the incoming audio, but the call is still active, when I talk to the other person he can hear me, but I can´t hear him, I'm sure about that because I once I have lost the incoming audio, I have called again that person, and he tells me that he was hearing me, also that he was talking to me, but I couldn´t hear him.

****** ADDITIONAL INFORMATION ******

I have used the following command: rtp debug ip 172.16.3.71

Where 172.16.3.71 is the IP of my laptop, where I'm using a softphone with a sip extension.

I have set up rtp ports from 8000 up to 20000

I'm connecting to my sip trunk provider through a dedicated network interface, he has assigned me the ip 172.21.0.26, I'm running a class B local network 172.16.0.0 and 172.19.0.0


Sent RTP packet to      172.16.3.71:8000 (type 00, seq 044705, ts 3211587232, len 000160)
Got  RTP packet from    172.16.3.71:8000 (type 00, seq 019951, ts 2152477314, len 000160)
Sent RTP packet to      172.16.3.71:8000 (type 00, seq 044706, ts 3211587392, len 000160)
Got  RTP packet from    172.16.3.71:8000 (type 00, seq 019952, ts 2152477474, len 000160)
Sent RTP packet to      172.16.3.71:8000 (type 00, seq 044707, ts 3211587552, len 000160)
Got  RTP packet from    172.16.3.71:8000 (type 00, seq 019953, ts 2152477634, len 000160)
Sent RTP packet to      172.16.3.71:8000 (type 00, seq 044708, ts 3211587712, len 000160)
Got  RTP packet from    172.16.3.71:8000 (type 00, seq 019954, ts 2152477794, len 000160)
Got  RTP packet from    172.16.3.71:8000 (type 00, seq 019955, ts 2152477954, len 000160)
Got  RTP packet from    172.16.3.71:8000 (type 00, seq 019956, ts 2152478114, len 000160)
Got  RTP packet from    172.16.3.71:8000 (type 00, seq 019957, ts 2152478274, len 000160)
Got  RTP packet from    172.16.3.71:8000 (type 00, seq 019958, ts 2152478434, len 000160)
Got  RTP packet from    172.16.3.71:8000 (type 00, seq 019959, ts 2152478594, len 000160)
Got  RTP packet from    172.16.3.71:8000 (type 00, seq 019960, ts 2152478754, len 000160)


Examining the full, I have found the following messages:

DEBUG[25651] rtp.c: Forcing Marker bit, because SSRC has changed
DEBUG[21916] rtp.c: Ignoring RTP 2833 Event: 000000e4. Not a DTMF Digit.
NOTICE[21946] rtp.c: Unknown RTP codec 98 received from '172.22.0.43'
DEBUG[7842] rtp.c: - RTP 2833 Event: 0000000a (len = 4)
VERBOSE[5162] logger.c:   == RTP Allocating from port range 8000 -> 20000
Comments:By: Alfredo Zambrano (azambrano) 2008-09-17 17:19:54

I forgot to tell, that I loss the incoming audio after any time (1 minute, or 2 minutes, there is no way to stablish an excat pattern), once audio is lost, it never comes again
By: Leif Madsen (lmadsen) 2008-09-17 17:39:33

This is most likely due to a configuration issue in sip.conf. Be sure to disable re-invites and the like which will cause audio to be redirected away from your Asterisk box, which will not work correctly when the system is behind NAT (or your devices are behind NAT).

Please utilize the public help channels such as the asterisk-users mailing list, or the #asterisk IRC channel on the Freenode network at irc.freenode.net

Thanks for using Asterisk!