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

[Andres]

>>
>>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?
>>    
>>
Use tcpdump to capture to a file both call scenarios.  Then use 
Wireshark to open the file.  You can then do an 'RTP-> Show All Streams' 
Analysis of the calls.  That alone would reveal whether the Audio is 
really there or not.  You can export that G711 Payload and listen to it 
with the Windows Media Player.

If you don't see the RTP in one direction then you might have a 
signalling problem.

Andres
http://www.neuroredes.com

>>Thanks,
>>
>>-Philip
>>  
>>    
>>
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