Use version 1.4 and samuel vinsons legacy device patch. You *could* use v0.9, but honestly your better up keeping up with current versions. Samuels patch MIGHT work with 1.5, but I havent tested this yet. Regards, Shayne. On 09/01/2010, at 12:10 AM, Chee Jin Li wrote: > Hi > > I have cross compiled PJSIP 1.0.3 for iPhone 3.1.2 according to Siphon guide at http://code.google.com/p/siphon/wiki/CompilationForiPhoneOS2_X . > > I have written a simple user agent to connect to the SIP server and the connection is successful, but when accepting incoming call the audio connection seem not able to connect to iPhone built-in microphone or speaker. Below is the console log file, at 23:59:32:904 the Port 1 is connected to Port 0 but stating Null Device: > > 23:59:32.901 UserAgent Incoming Call from 0 > 23:59:32.902 strm0x859d74 VAD temporarily disabled > 23:59:32.902 strm0x859d74 Encoder stream started > 23:59:32.903 strm0x859d74 Decoder stream started > 23:59:32.903 pjsua_media.c Media updates, stream #0: GSM (sendrecv) > 23:59:32.903 pjsua_media.c pjsua_set_snd_dev(): attempting to open devices @16000 Hz > 23:59:32.903 ec0x2390e0 Echo suppressor created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=20 ms > 23:59:32.904 conference.c Port 1 (sip:100 at 192.168.0.8:5060) transmitting to port 0 (Null Device) > 23:59:32.904 conference.c Port 0 (Null Device) transmitting to port 1 (sip:100 at 192.168.0.8:5060) > 23:59:32.904 pjsua_core.c TX 806 bytes Response msg 200/INVITE/cseq=1 (tdta0x856a00) to UDP 192.168.0.8:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bK-d8754z-7d5aa01c01789251-1---d8754z- > Call-ID: M2Q5YmExMzQyYTIwMjExNDM4OTBlNWU3ODcxNWFmM2U. > From: "Jin Li Chee" <sip:100@192.168.0.8>;tag=bf48e02a > To: <sip:101 at 192.168.0.8>;tag=OmIL-dI3-fV9Q0Ta4mn0e6csiJvsAqk1 > CSeq: 1 INVITE > Contact: <sip:101 at 192.168.0.199:5060> > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS > Supported: replaces, 100rel, norefersub > Content-Type: application/sdp > Content-Length: 253 > > v=0 > o=- 3471955172 3471955173 IN IP4 192.168.0.199 > s=pjmedia > c=IN IP4 192.168.0.199 > t=0 0 > a=X-nat:0 > m=audio 4000 RTP/AVP 3 101 > a=rtcp:4001 IN IP4 192.168.0.199 > a=rtpmap:3 GSM/8000 > a=sendrecv > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > --end msg-- > 23:59:32.906 UserAgent Call 0 State = OK > 23:59:33.012 pjsua_core.c RX 434 bytes Request msg ACK/cseq=1 (rdata0x832064) from UDP 192.168.0.8:5060: > ACK sip:101 at 192.168.0.199:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-d8754z-f11b32342f046353-1---d8754z-;rport > Max-Forwards: 70 > Contact: <sip:100 at 192.168.0.8:5060> > To: <sip:101 at 192.168.0.8>;tag=OmIL-dI3-fV9Q0Ta4mn0e6csiJvsAqk1 > From: "Jin Li Chee"<sip:100@192.168.0.8>;tag=bf48e02a > Call-ID: M2Q5YmExMzQyYTIwMjExNDM4OTBlNWU3ODcxNWFmM2U. > CSeq: 1 ACK > User-Agent: 3CXPhoneSystem 7.1.7060.0 > Content-Length: 0 > > > --end msg-- > 23:59:33.012 UserAgent Call 0 State = OK > > Is anything I miss out during the cross-compile? Or is it I should add something during my user agent compilation to make it to work with the iPhone Sound Audio Queue? > > Thanks in advance. > Jin Li Chee > > > > > _______________________________________________ > 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 =================================== Shayne O'Neill Development Mobile, Web and Business process integration. shayne.oneill at gmail.com 0400247091 Ask me about how Alfresco can help your business grow.
