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
