[sip-comm] JMF stops playing


#1

Hi,

i have some problems getting the SIP-Communicator to fly.
I can establish a call without any problems (via asterisk or ser), but when the audio session is established (sc<>sc or sc<>x-lite>) the audio stops after 3 seconds. Audio still runs in a single direction depending on which side initiated the call. There is no NAT involved here.

Any ideas?

I have started SIP-Communicator via Java Web Start and via Eclipse (build from src) on a w2k machine.

Probably JMF freaks out...

···

--
Simon Schoar
Bell Labs Europe
Advanced Technologies

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#2

Hello Simon,

It might be JMF or it might be the sip-communicator. Look through the logs and see what happens. It is possible that the stream has just been interrupted and then renewed without the sip-communicator taking account of the new "stream received event" This should normally be easy to spot in the log files. Can you have a look?

We'll be soon having an asterisk here and we'll be doing some compatibility tests. Until then however you'll have to be staring at those long logs in order to help us get it fixed:) (of course this is all highly appreciated).

Cheers
Emil

Schoar, Simon (Simon)** CTR ** wrote:

···

Hi,

i have some problems getting the SIP-Communicator to fly.
I can establish a call without any problems (via asterisk or ser), but when the audio session is established (sc<>sc or sc<>x-lite>) the audio stops after 3 seconds. Audio still runs in a single direction depending on which side initiated the call. There is no NAT involved here.

Any ideas?

I have started SIP-Communicator via Java Web Start and via Eclipse (build from src) on a w2k machine.

Probably JMF freaks out...

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