Debugging RTP packets sending

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can you elaborate into how u setup the sending address?
from what i have seen, you need to setup the transport address before
sending the frames, maybe this is wrong some way.
also, u may have a problem with nat related issues, does your server uses
mediaproxy?
also, when you use regular pjsua transport is it working?


On Wed, Mar 24, 2010 at 5:31 PM, Fabio Pietrosanti (naif) <
lists at infosecurity.ch> wrote:

> 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
>>>
>>>
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>>>
>>>
>>
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>>
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>>
>>
>
>
> _______________________________________________
> 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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