[Serusers] Voice loss after establishing call

[Jan Janak]

The phones are probably configured to use silence-suppression. That
means they do not send any media packet if the audio level is below some
threshold. This unfortunately does not work very well with NAT
traversal, because the RTP proxy has to receive at least one packet from
both sides in order to work properly.

Try to disable silence suppression.

  Jan.

On 01-05 16:05, Arnd Vehling wrote:
> Hello,
> 
> we currently experience "incoming voice loss" after an outgoing
> call has been established. i.e. after the remote party picks up the
> call you dont hear anything for 1-2 seconds which results in both
> sides of the connection saying repeatedly "Hello" until both sides
> can hear each other.
> 
> This phenomenon occurs with nated and non natted clients so it
> doesnt seem to be related to "rtpproxy" or other NAT problems.
> 
> It occurs between  SIP<>SIP and SIP<>PSTN Calls (via a cisco gw) 
> independantly of the UA type used.
> 
> Has anyone an idea where to look for the bug/where to start debugging?
> 
> best regards,
> 
>   Arnd
> 
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