20 Apr
2007
20 Apr
'07
8:24 a.m.
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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