Debugging RTP packets sending

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Unfortunately it's not possible to use wireshark.

Packets are directly sent to GPRS wan interface of a Nokia S60 phone 
runing Symbian OS where no single network analyzer exists.
My problems exists only in GPRS and UMTS  but not in Wifi (that's the 
only environment where i can use a network analyzer).

So i need a way to be able to see all the RTP related activity of PJSIP 
in order to debug the condition from PJSIP logs as the mobile operating 
system does not permit to use a network analyzer.

Fabio

On 24/03/10 15.50, Peter Lukac wrote:
> Hi
> Try wireshark at http://www.wireshark.org
> it's very good multiplatform network analyzer and is free. Contains support
> for SIP, RTP and lot of another protocols...
>
> regards
>
>    
>> Hi all,
>>
>> i am struggling with debugging a one-way audio situation.
>>
>> I analyzed everything related to SIP and SDP attributes and everything
>> seems fine.
>>
>> I look at PJSIP logs in realtime during debugging.
>>
>> Particulary i am interested in:
>> - see every RTP packets sent out
>> - see to which address/port the RTP packets are sent out
>>
>> Is there a way to check RTP related information by enabling some macro
>> and/or define?
>>
>> I am already running with PJSIP at maximum debug level.
>>
>> Fabio
>>
>>
>> _______________________________________________
>> Visit our blog: http://blog.pjsip.org
>>
>> pjsip mailing list
>> pjsip at lists.pjsip.org
>> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>>      
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip at lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>    




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