[Serusers] Help: I can't hear audio although I receive the RTP packets

[Victor Huertas Garcia]

Firt of all, thank you very much for your answer.

I tell you the rarest. With NAT enabled, I call a normal endpoint (SIP 
softphone) in located in a PC and it works fine. Audio is heared in both 
endpoints. But when I call to PSTN through a VoIP GW it happens what I told 
in the last e-mail. I can see how the RTP packets trespasses the NAT and 
reach the enpoint behind the NAT (exactly the same as with the normal 
endpoint) but no audio is heard. What's more, no ICMPs are generated by the 
PC so I interprehet that the destination RTP port is correct. I have tried 
it with three different endpoints (Cisco SIP phone, and two different 
softphones, one of both is Eyebeam) with the same result.

That is what is misleading me the most... Why does it work if I call to a 
normal endpoint? If it is NAT fault, it should happen the same as when I 
call to PSTN.

Obviously that NAT is somehow responsible for this problem but I can't 
figure out what's going on. I have compared with Ethereal every sigle packet 
for calls to normal endpoint and VoIP GW and all the translations in SDP are 
preformed identically.

I just wonder if there has been cases in which the RTP reaches and endpoint 
(with no ICMPs at all, remember) and no sound is heared.

Thanks in advance.

Victor
>From: "Greger V. Teigre" <greger at teigre.com>
>To: Victor Huertas Garcia <vhuertas at hotmail.com>
>CC: serusers at iptel.org
>Subject: Re: [Serusers] Help: I can't hear audio although I receive the RTP 
>packets
>Date: Fri, 20 Apr 2007 08:24:54 +0200
>
>Think in straight lines: If you remove the NAT and it works, then the NAT 
>is the problem. It also means that if both RTP and SIP signalling looks the 
>same in both cases, you have either overlooked a difference, or they are 
>really the same, but the NAT is doing something and your tests haven't 
>revealed it.
>
>So, have you captured signalling and RTP on both end-points? So you know 
>the NAT is not dropping RTP?
>If you see streams both ways on BOTH end-points, then I would say the NAT 
>either garbles SIP messages (and you have overlooked something) or it 
>garbles RTP (which in case you should probably see something in the user 
>agents' error logs?)
>g-)
>
>Victor Huertas Garcia wrote:
>>
>>
>>
>>Hello all,
>>
>>My problem is one of the most strange problems I have ever found.
>>I am using a SER (v0.8.14) configured to support Nated endpoints.
>>I make a call from a NATed SIP endpoint towards another endpoint which is 
>>not NATed (static NAT IP to IP translation).
>>Everything seems to work well, as all the mechanism to solve NAT are 
>>triggered and all the packets (SDP part) are correctly translated by the 
>>SER.
>>I captured all the SIP packets and the call is correctly established. I 
>>can see how the NATed endpoint send and receive RTP packets with no 
>>problem (aparently). Even all RTP ports are correct according to the 
>>previous negotiation (SDP) between both endpoints. Moreover, the 
>>negotiated codecs are fully supported.
>>
>>What is happening is REALLY rare... I CAN'T HEAR ANYTHING AT ALL.
>>
>>All audio is well configured and enable. What's more, as soon as I take 
>>out the NAT, I make the same call to the same destination and voila! RTP 
>>packets are received and HEARD!!! The voice codecs used for communications 
>>are exactly the same as the ones used with NAT.
>>
>>I have checked every single packet and SIP message and everything seems 
>>OK... I don't what I can do.
>>
>>Any idea? I have packet captures. Anyone who wants to have a look, please 
>>tell me.
>>
>>Thank you very much for your attention.
>>
>>Victor
>>
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>>

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