[SIPForum-discussion] Regarding RTP audio packets getting dropped by the media gateway with SIP calls.

[Saurabh Shah]

Hi Badri, 


pTime is a declarative parameter in SDP. In absent of pTime, default pTime value of negotiated codec must be used. 
In your case, pTime is not negotiated and hence 20mSec (default for PCMU) must be used. 
But 1.72 is sending @ an interval of 60mSec. 
This could be the reason. 


Regards, 
Saurabh Shah 

----- Original Message -----

From: "Badri Ranganathan" <badri at arcatech.com> 
To: discussion at sipforum.org 
Sent: Wednesday, April 23, 2014 1:52:09 PM 
Subject: [SIPForum-discussion] Regarding RTP audio packets getting dropped by the media gateway with SIP calls. 



Hi all, 

I have 2 SIP endpoints (192.168.1.76 making the call and 192.168.1.72 answering the call). The call is made via a proxy server in between which is 195.186.130.4. 
Then 192.168.1.76 plays out audio RTP and 192.168.1.72 is supposed to record this audio. That is the test. 

I am facing a strange problem with RTP audio not being received by 1.72 although it is being played out from 1.76 

The calls are established through SIP without any problem. The SDP exchange also seems to be fine with both ends exchanging PCMU (Owner in SDP is 195.186.130.4 and media connection endpoint is 195.186.130.132 (presume it is some media gateway assigned by the proxy)) 

The endpoint 192.168.1.76 has played out audio RTP but the receiver has not received anything. (AC_rec.txt will confirm that) 
However I am able to see that the RTCP exchange between these endpoints through the media gateway is fine without any issue. 

I cannot view or debug at the media gateway since I have no authority to do it. But my reason to believe that the RTP must have reached the media gateway and that it is the media gateway dropping it – The RTCP packets are getting across without a problem !!! Therefore the RTP packets must have definitely reached the MG and the MG is dropping it because it doesn’t like them for some reason L 

Why does the media gateway not forward the RTP packets that 192.168.1.76 has sent? 
Any way of debugging through the RTCP protocol itself? 
Is there any issue in the SDP negotiation between the SIP endpoints or with the RTP packets itself? 

Will be very helpful if someone can please look at the wireshark traces attached with this email. Any help in this regard is very much appreciated. 

Thanks, 
Badri. 


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