[asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?

[Sergio]

A similar issue happens to us.
Make sure that, for inbound AND outbound calls rtp packets are reaching the 
other endpoint.
If a NAT device(s) is between the endpoints make sure that the device NATs the 
traffic on BOTH ways (inbound AND outbound).

Regards

On Saturday 27 September 2008 23:54:37 Philip Prindeville wrote:
> I've got the following situation.  I'm running Asterisk 1.4.18 on a
> firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones
> behind it.
>
> I'm peering SIP with a Coppercom switch sitting behind an SBC.
>
> On outbound calls, I get 2-way voice, no worries.
>
> On inbound calls, I get one-way voice (I can hear the caller but they
> can't hear me).
>
> I've looked at tcpdumps of the RTP traffic, and the addresses and port
> numbers correspond to what's in the SIP INVITE/OK messages (assuming
> that they don't somehow get munged by NAT after tcpdump looks at them --
> there is no NAT device upstream of my Asterisk firewall).
>
> I'll look into using Record() or Monitor() to capture the phone call,
> but if there's any conversion being done by codecs then that won't
> eliminate the possibility that the code itself is misconfigured or buggy
> and generating a bad stream on one of the legs...
>
> Anyone have an idea about how to best go about troubleshooting this?
>
> Thanks,
>
> -Philip
>
>
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