Hi Benny, have looked at the traces I sent? We looked with Aastra engineers at the trace and they saw that RTCP seams to come early. We tested on several computers and saw that it sometimes works. It looks like a threading/timing problem in pjsip? The hardware phones are Aastra IP-Phones sending media directly to PC with pjmedia stack. Regards Michael. --- On Fri, 1/29/10, Michael <michael_zurich at yahoo.com> wrote: > From: Michael <michael_zurich@xxxxxxxxx> > Subject: Re: No Audio with Aastra PBX > To: "pjsip list" <pjsip at lists.pjsip.org> > Date: Friday, January 29, 2010, 3:26 PM > Hi Paulo > Hi Benny > > Attached 2 files from same computer. 2 Calls one works with > pjsip one not. Please note that both call sources are > working with other sip client not using pjsip. > > Regards & Thanks > > Michael > > --- On Thu, 1/21/10, Paulo Rog?rio Panhoto <paulo at voicetechnology.com.br> > wrote: > > > From: Paulo Rog?rio Panhoto <paulo@xxxxxxxxxxxxxxxxxxxxxx> > > Subject: Re: No Audio with Aastra PBX > > To: "pjsip list" <pjsip at lists.pjsip.org> > > Date: Thursday, January 21, 2010, 2:05 PM > > Hi Michael, > > > > Depending on the PBX configuration (most PBX have > this > > setting), the RTP might be established between the > terminals > > directly or through the PBX. I thought it could be > possible > > the softphone cannot find a network route to the > Aastra > > telephone. But, once it works with phoner, there is a > > network path anyway. > > > > > > If you send one capture file for PJSUA and one for > phoner > > (containing SIP message flow with SDP, which contains > the > > RTP negotiation details), it might possible to figure > out > > what is going wrong with your environment. > > > > > > Regards, > > > > Paulo. > > > > 2010/1/21 Benny Prijono <bennylp at teluu.com> > > > > On Thu, Jan 21, 2010 at 6:39 AM, Michael > > <michael_zurich at yahoo.com> > > wrote: > > > > > Hi Benny > > > > > Hi Paulo > > > > > > > > > > Thanks for the info. > > > > > > > > > > I checking if RTP Packets are received with > > pjsua's dq (dump quality of current call). No RTP > packet > > is received. Please take care about the following: > > > > > > > > > > 2 cases: > > > > > > > > > > Case 1: Call from ISDN trunk over Aastra PBX > works > > fine. All audio and packets. > > > > > > > > > > Case 2: Call from an internal Aastra hardware > phone to > > pjsua does signal and connect and on Aastra phone you > hear > > the pjsua client but on pjsua client you receive no > packets > > but data arrives. No Router involved, PBX is directly > > connected over a switch. PBX ip 172.16.4.1, pjsua ip > > 172.16.2.12 net mask 255.255.0.0. But note, it works > fine > > with phoner software client. There must be an issue in > pjsua > > (pj...) because it works with other software clients. > > > > > > > > > > > > > > > If we follow your logic, pjsua also works fine with > > other software, so > > > > there must be an issue with the PBX. > > > > > > > > > Could it be that Aastra transmits media on a port > or > > mode that pjsua can't read? > > > > > > > > > > > > > It could be, but we can't never be sure, until > > you follow our > > > > suggestions and find out what's wrong. > > > > > > > > Cheers > > > > ?Benny > > > > > > > > _______________________________________________ > > > > 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 > > > > > > > > > > -----Inline Attachment Follows----- > > > > _______________________________________________ > > 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 > > > > > ? ? ?
