2 May
2004
2 May
'04
12:05 a.m.
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
6 May
6 May
10:51 p.m.
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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Arnd Vehling -
Jan Janak