[asterisk-users] No audio on remote SIP calls

[Benny Amorsen]

Jonathan Moore <supermegatron at gmail.com> writes:

> The idea of RTP being to blame would make sense though.  I can
> still transfer and such, and watching the console, I see when I press
> various keys on the phone, so it seems that the SIP traffic is working
> out fine.  (I do understand that right?  SIP == control RTP == voice
> in a very generic sense?)

> I plan to take a packet trace in the morning on the asterisk server and
> see what is going on at that level.  Hints as to what I should be looking
> for?

Start by looking at pure SIP traffic by doing -s0 -v and filtering on port
5060. Notice the media streams being negotiated, and look at the IP
addresses and ports.

If that doesn't help, remove the port 5060 filter and look again at the
raw traffic -- but that can be a lot of traffic.

My guess: You have STUN enabled on the phones.


/Benny