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

[Philip Prindeville]

Andres wrote:
>>> 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.
>   

I'm running wireshark 1.0.3.  I've opened the captures...  How do I 
examine the streams?  I don't follow what you're saying above.

And does anyone have a plugin that would allow actual playback of the 
.pcap files' audio packets?

Thanks,

-Philip

> If you don't see the RTP in one direction then you might have a 
> signalling problem.
>
> Andres
> http://www.neuroredes.com
>
>   
>>> Thanks,
>>>
>>> -Philip
>>>  
>>>    
>>>
>>>