Hi Fabio, we have done a lot of test with PJSIP in mobile environment and there are a couple of things you can do: 1. Fisrt are you sure the network doesn't block ports etc, I'v seen that a lot of operators block traffic perhaps you could try running PJSIP on a PC with a mobile connection and in that case you can use wireshark, just to make sure you have no network issues. 2. you could add an object similar to SRTP object that copies RTP packets to file. The I'm just querious how you should get PJSIP work on GPRS, all the test we have done shows this is very hard in rel conditions even if you send multiple packets per frame, especially on the uplink. BR/Olle _____ From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org] On Behalf Of Fabio Pietrosanti (naif) Sent: den 24 mars 2010 17:04 To: pjsip list Subject: Re: Debugging RTP packets sending In my situation i use FreeSWITCH.org that make by default RTP media relay by dynamically discovering if the peer is behind nat or not. I also force FS to act as a proxy for the media. Now something it's not working, FS does not receive RTP packets of one leg. I need to find out what PJSIP is sending out to the FS (that is not receiving anything), also to understand if PJSIP is sending something and if it's sending something if the target IP/PORT is correct (on SDP attribute it seems correct). Fabio On 24/03/10 16.56, nir elkayam wrote: if it one way then this might be the solution. u can't usually have one way connection when both sides are behaind symetric nat. this is because the public address of the other side is never reveled when he doesn't sends any packets.. On Wed, Mar 24, 2010 at 4:37 PM, Fabio Pietrosanti (naif) <lists at infosecurity.ch> wrote: 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 -- ??? ?????? ??: 050-3930056 nir.elkayam at gmail.com _______________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100324/ce8649db/attachment.html>
