Hi Tony,
Can you get an ethereal trace of the RTP packets on both directions? A short analysis of those streams (from within the ethereal tools) would help us find the problem.
Michael.
Tony Mountifield wrote:
I'm using asterisk-oh323-0.6.5 with the Janus patch 4 versions of pwlib (v1.6.6.3) and openh323 (v1.13.5.3), and using it to connect to my provider's switch.
The effect that I am seeing is that a call starts off fine, but suddenly after a few minutes the audio coming into Asterisk via OH323 gets very broken up to the point of being about 90% silence with occasional brief snippets of audio getting through.
When this happens, the audio going out from Asterisk to the other end is still fine, with no disturbances.
I have observed this both when using SIP for the local leg of the call and when using IAX.
I'm not really sure where to look to diagnose this, not whether it is likely to be an Asterisk problem or something in the switch.
Any advice would be appreciated!
Cheers Tony
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