Hi Benny,Olivier I had a similar problem. It looks like PJSIP uses the Internet IP address(obtained from the STUN server) in the contact header for calls relating to a registered account. If you don't have a local LAN account with a contact override it also uses the public IP address when receiving local LAN calls. I think some routers may handle this with a loop back and some not. The one UA actually remains in the calling state while the other is confirmed. Sound does go through but when the UA in calling state times out the call will get disconnected. It takes long to disconnect due to the fact that it retries 10 times (due to wrong contact) and finally disconnects when it gives up. Benny, it would be useful to be able to comment a config file with something like -comment= example config with local account override --stun-srv=stun.fwdnet.net:3478 --comment=Following is the local lan account with a override - remove this line in app. --id=sip:10.0.0.10 --contact=sip:10.0.0.10 --next-account --comment=Following is the local registered account - remove this line in app. --registrar=sip:fwd.pulver.com:5060 --id=sip:8664xx:xxxxxx at fwd.pulver.com --proxy=sip:fwd.pulver.com:5060 --realm=fwd.pulver.com --username=8664xx --password=xxxxxx regards Lafras ----- Original Message ----- From: "Benny Prijono" <bennylp@xxxxxxxxx> To: "pjsip embedded/DSP SIP discussion" <pjsip at lists.pjsip.org> Sent: Wednesday, October 10, 2007 6:51 PM Subject: Re: [pjsip] pjsip for WM5/C# > Hi Olivier, > > Olivier Beytrison wrote: > > I still experience some issues : > > - When calling pjsua_acc_del(accountID), the on_reg_state() callback > > isn't called. Looking at the logs, pjsua delete the account before > > receiving the 200 OK from the server, this preventing it from calling > > the on_reg_state(pjsua_acc_id acc_id) callback, because the account > > have already been deleted > > I think that's the expected behavior. From application's view, the > deletion happens immediately. Once the account is deleted, it is > gone as far as application is concerned, thus it will not hear any > more callbacks called on behalf of the account. > > > - During a call, some times It doesn't send any sound. After putting on > > hold the call, and retrieving it, the sounds works. And i get a lot > > of BAD Ptr for the RTP session > > Not sure about this. Maybe you can get more information by following > the troubleshooting procedure from the sound problem Wiki > (http://www.pjsip.org/trac/wiki). > > > - when calling pjsua_call_hangup(), it takes most of the time 8 to 15 > > seconds for the call to hang up. Same goes when the other UA initiate > > the hang up > > This sounds like problem with Contact header calculation. Maybe the > IP address that is selected in Contact header is not the one that's > reachable from remote UA. > > > About the IMS-Specific things for SIP, it is mainly Headers. (like the > > P-Access-Info header which describe which kind of "connection" the > > signaling is sent on, like 3gpp-umts-cell, wireless-network, ect ect). I > > can give you a small overview of the 3gpp-specific headers if you're > > interested. > > An IMS overview will be great! Not that I'm planning to implement > them (just yet), but any free info is always good. :) > > cheers, > -benny > > > That's it for now, work is going on for the GUI, and hopefully we should > > have a beta version in about 1 week. > > > > Regards, > > > > Olivier B. > > > > > -- > Benny Prijono > http://www.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 >
