[pjsip] pjsua transfer problem.

Benny Prijono bennylp at pjsip.org
Thu Nov 22 00:32:00 EST 2007


Hi,

there was a suspicious failure when we receive 200/OK for INVITE:

 >  10:49:26.931   pjsua_call.c SDP negotiation has failed: No 
active media stream after negotiation (PJMEDIA_SDPNEG_ENOMEDIA) 
[status=220048]

I'm not sure why the SDP was rejected, but one thing for sure, 
you're using a pretty old pjsip, so maybe new version will work 
better. Can you try with using pjsua application from the 0.8 version?

cheers,
  -benny

boujbel zied wrote:
> Hi,
> I'm working on  application that should call a phone A and transfer the call to B (with pjsua_call_xfer).
> the transfer it did well but there is no communication between the two phones (cann't hear neither talk).
> Note that I deal with an Asterisk Server.
> Any idea is welcome.
> thanks
> 
>  10:49:25.197 os_core_unix.c pjlib 0.5.10.4 for POSIX initialized
>  10:49:25.197 sip_endpoint.c Creating endpoint instance...
>  10:49:25.199          pjlib select() I/O Queue created (0x811b66c)
>  10:49:25.200 sip_endpoint.c Module "mod-msg-print" registered
>  10:49:25.200 sip_transport. Transport manager created.
>  10:49:25.200 sip_endpoint.c Module "mod-pjsua-log" registered
>  10:49:25.200 sip_endpoint.c Module "mod-tsx-layer" registered
>  10:49:25.200 sip_endpoint.c Module "mod-stateful-util" registered
>  10:49:25.200 sip_endpoint.c Module "mod-ua" registered
>  10:49:25.201 sip_endpoint.c Module "mod-pjsua" registered
>  10:49:25.201 sip_endpoint.c Module "mod-invite" registered
>  10:49:25.271      pasound.c PortAudio sound library initialized, status=0
>  10:49:25.271      pasound.c PortAudio host api count=1
>  10:49:25.271      pasound.c Sound device count=1
>  10:49:25.272          pjlib select() I/O Queue created (0x8153f24)
>  10:49:25.273 sip_endpoint.c Module "mod-evsub" registered
>  10:49:25.274 sip_endpoint.c Module "mod-presence" registered
>  10:49:25.274 sip_endpoint.c Module "mod-refer" registered
>  10:49:25.274 sip_endpoint.c Module "mod-pjsua-pres" registered
>  10:49:25.274 sip_endpoint.c Module "mod-pjsua-im" registered
>  10:49:25.274 sip_endpoint.c Module "mod-pjsua-options" registered
>  10:49:25.274   pjsua_core.c 1 SIP worker threads created
>  10:49:25.274   pjsua_core.c pjsua version 0.5.10.4 for i686-pc-linux-gnu initialized
>  10:49:25.275   pjsua_core.c SIP UDP socket reachable at 192.168.1.213:50060
>  10:49:25.275   udp0x81734cc SIP UDP transport started, published address is 192.168.1.213:50060
>  10:49:25.276    pjsua_acc.c Account SIP:131 at localhost added with id 0
>  10:49:25.276   pjsua_core.c TX 350 bytes Request msg REGISTER/cseq=64678 (tdta0x81750f4) to UDP 127.0.0.1:5060:
> REGISTER sip:localhost SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000367458b6bc623
> Max-Forwards: 70
> From: ;tag=25e0000267458b6bc623
> To: 
> Call-ID: 25e0000167458b6bc623
> CSeq: 64678 REGISTER
> Contact: 
> Expires: 55
> Content-Length:  0
> 
> 
> --end msg--
>  10:49:25.276    pjsua_acc.c Registration sent
>  10:49:25.276 sip_endpoint.c Module "mod-handle-requests" registered
>  10:49:25.276   pjsua_call.c Making call with acc #0 to SIP:131 at 192.168.1.10
>  10:49:25.277   pjsua_core.c TX 900 bytes Request msg INVITE/cseq=1714636915 (tdta0x817833c) to UDP 192.168.1.10:5060:
> INVITE sip:131 at 192.168.1.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000667458b6bc623
> Max-Forwards: 70
> From: sip:131 at localhost;tag=25e0000467458b6bc623
> To: sip:131 at 192.168.1.10
> Contact: 
> Call-ID: 25e0000567458b6bc623
> CSeq: 1714636915 INVITE
> Allow: INVITE, ACK, BYE, CANCEL, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS
> Supported: replaces, norefersub
> Content-Type: application/sdp
> Content-Length:   403
> 
> v=0
> o=- 3404648965 3404648965 IN IP4 0.0.0.0
> s=pjmedia
> c=IN IP4 0.0.0.0
> t=0 0
> m=audio 0 RTP/AVP 103 102 104 117 3 0 8 101
> a=rtcp:0 IN IP4 0.0.0.0
> a=rtpmap:103 speex/16000
> a=rtpmap:102 speex/8000
> a=rtpmap:104 speex/32000
> a=rtpmap:117 iLBC/8000
> a=fmtp:117 mode=20
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=sendrecv
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> 
> --end msg--
>  10:49:25.277       makeCall -- Call 0 state=CALLING
>  10:49:25.278   pjsua_core.c RX 514 bytes Response msg 407/INVITE/cseq=1714636915 (rdata0x81738dc) from UDP 192.168.1.10:5060:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 192.168.1.213:50060;branch=z9hG4bKPj25e0000667458b6bc623;received=192.168.1.213;rport=50060
> From: sip:131 at localhost;tag=25e0000467458b6bc623
> To: sip:131 at 192.168.1.10;tag=as785e783b
> Call-ID: 25e0000567458b6bc623
> CSeq: 1714636915 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63da021a"
> Content-Length: 0
> 
> 
> --end msg--
>  10:49:25.278   pjsua_core.c TX 299 bytes Request msg ACK/cseq=1714636915 (tdta0x817aaf4) to UDP 192.168.1.10:5060:
> ACK sip:131 at 192.168.1.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000667458b6bc623
> Max-Forwards: 70
> From: sip:131 at localhost;tag=25e0000467458b6bc623
> To: sip:131 at 192.168.1.10;tag=as785e783b
> Call-ID: 25e0000567458b6bc623
> CSeq: 1714636915 ACK
> Content-Length:  0
> 
> 
> --end msg--
>  10:49:25.278   pjsua_core.c TX 1068 bytes Request msg INVITE/cseq=1714636916 (tdta0x817833c) to UDP 192.168.1.10:5060:
> INVITE sip:131 at 192.168.1.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000767458b6bc623
> Max-Forwards: 70
> From: sip:131 at localhost;tag=25e0000467458b6bc623
> To: sip:131 at 192.168.1.10
> Contact: 
> Call-ID: 25e0000567458b6bc623
> CSeq: 1714636916 INVITE
> Allow: INVITE, ACK, BYE, CANCEL, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS
> Supported: replaces, norefersub
> Proxy-Authorization: Digest username="131", realm="asterisk", nonce="63da021a", uri="sip:131 at 192.168.1.10", response="a91966d158535db8463ac523fe353e99", algorithm=md5
> Content-Type: application/sdp
> Content-Length:   403
> 
> v=0
> o=- 3404648965 3404648965 IN IP4 0.0.0.0
> s=pjmedia
> c=IN IP4 0.0.0.0
> t=0 0
> m=audio 0 RTP/AVP 103 102 104 117 3 0 8 101
> a=rtcp:0 IN IP4 0.0.0.0
> a=rtpmap:103 speex/16000
> a=rtpmap:102 speex/8000
> a=rtpmap:104 speex/32000
> a=rtpmap:117 iLBC/8000
> a=fmtp:117 mode=20
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=sendrecv
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> 
> --end msg--
>  10:49:25.279       makeCall -- Call 0 state=CALLING
>  10:49:25.280   pjsua_core.c RX 431 bytes Response msg 100/INVITE/cseq=1714636916 (rdata0x81738dc) from UDP 192.168.1.10:5060:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.1.213:50060;branch=z9hG4bKPj25e0000767458b6bc623;received=192.168.1.213;rport=50060
> From: sip:131 at localhost;tag=25e0000467458b6bc623
> To: sip:131 at 192.168.1.10
> Call-ID: 25e0000567458b6bc623
> CSeq: 1714636916 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: 
> Content-Length: 0
> 
> 
> --end msg--
>  10:49:25.432   pjsua_core.c RX 447 bytes Response msg 180/INVITE/cseq=1714636916 (rdata0x81738dc) from UDP 192.168.1.10:5060:
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 192.168.1.213:50060;branch=z9hG4bKPj25e0000767458b6bc623;received=192.168.1.213;rport=50060
> From: sip:131 at localhost;tag=25e0000467458b6bc623
> To: sip:131 at 192.168.1.10;tag=as2d7b4585
> Call-ID: 25e0000567458b6bc623
> CSeq: 1714636916 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: 
> Content-Length: 0
> 
> 
> --end msg--
>  10:49:25.432       makeCall -- Call 0 state=EARLY
>  10:49:25.779   pjsua_core.c TX 350 bytes Request msg REGISTER/cseq=64678 (tdta0x81750f4) to UDP 127.0.0.1:5060:
> REGISTER sip:localhost SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000367458b6bc623
> Max-Forwards: 70
> From: ;tag=25e0000267458b6bc623
> To: 
> Call-ID: 25e0000167458b6bc623
> CSeq: 64678 REGISTER
> Contact: 
> Expires: 55
> Content-Length:  0
> 
> 
> --end msg--
>  10:49:26.779   pjsua_core.c TX 350 bytes Request msg REGISTER/cseq=64678 (tdta0x81750f4) to UDP 127.0.0.1:5060:
> REGISTER sip:localhost SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000367458b6bc623
> Max-Forwards: 70
> From: ;tag=25e0000267458b6bc623
> To: 
> Call-ID: 25e0000167458b6bc623
> CSeq: 64678 REGISTER
> Contact: 
> Expires: 55
> Content-Length:  0
> 
> 
> --end msg--
>  10:49:26.931   pjsua_core.c RX 762 bytes Response msg 200/INVITE/cseq=1714636916 (rdata0x81738dc) from UDP 192.168.1.10:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.1.213:50060;branch=z9hG4bKPj25e0000767458b6bc623;received=192.168.1.213;rport=50060
> From: sip:131 at localhost;tag=25e0000467458b6bc623
> To: sip:131 at 192.168.1.10;tag=as2d7b4585
> Call-ID: 25e0000567458b6bc623
> CSeq: 1714636916 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: 
> Content-Type: application/sdp
> Content-Length: 287
> 
> v=0
> o=root 28818 28818 IN IP4 192.168.1.10
> s=session
> c=IN IP4 192.168.1.10
> t=0 0
> m=audio 12538 RTP/AVP 0 8 3 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 
> --end msg--
>  10:49:26.931       makeCall -- Call 0 state=CONNECTING
>  10:49:26.931   pjsua_call.c SDP negotiation has failed: No active media stream after negotiation (PJMEDIA_SDPNEG_ENOMEDIA) [status=220048]
>  10:49:26.931   pjsua_core.c TX 299 bytes Request msg ACK/cseq=1714636916 (tdta0x817dac4) to UDP 192.168.1.10:5060:
> ACK sip:131 at 192.168.1.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000867458b6bc623
> Max-Forwards: 70
> From: sip:131 at localhost;tag=25e0000467458b6bc623
> To: sip:131 at 192.168.1.10;tag=as2d7b4585
> Call-ID: 25e0000567458b6bc623
> CSeq: 1714636916 ACK
> Content-Length:  0
> 
> 
> --end msg--
>  10:49:26.931       makeCall -- Call 0 state=CONFIRMED
>  10:49:26.932   pjsua_core.c TX 524 bytes Request msg REFER/cseq=1714636917 (tdta0x817ded4) to UDP 192.168.1.10:5060:
> REFER sip:131 at 192.168.1.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000967458b6bc623
> Max-Forwards: 70
> From: sip:131 at localhost;tag=25e0000467458b6bc623
> To: sip:131 at 192.168.1.10;tag=as2d7b4585
> Contact: 
> Call-ID: 25e0000567458b6bc623
> CSeq: 1714636917 REFER
> Event: refer
> Expires: 300
> Accept: message/sipfrag;version=2.0
> Allow-Events: presence, refer
> Refer-To: SIP:5147729433 at 192.168.1.10
> Referred-By: SIP:131 at localhost
> Content-Length:  0
> 
> 
> --end msg--
>  10:49:26.932 evsub0x817da74 Subscription state changed NULL --> SENT
>  10:49:26.932   pjsua_core.c RX 447 bytes Response msg 202/REFER/cseq=1714636917 (rdata0x81738dc) from UDP 192.168.1.10:5060:
> SIP/2.0 202 Accepted
> Via: SIP/2.0/UDP 192.168.1.213:50060;branch=z9hG4bKPj25e0000967458b6bc623;received=192.168.1.213;rport=50060
> From: sip:131 at localhost;tag=25e0000467458b6bc623
> To: sip:131 at 192.168.1.10;tag=as2d7b4585
> Call-ID: 25e0000567458b6bc623
> CSeq: 1714636917 REFER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: 
> Content-Length: 0
> 
> 
> --end msg--
>  10:49:26.932 evsub0x817da74 Subscription state changed SENT --> ACCEPTED
>  10:49:26.933       makeCall Call 0: transfer status=100 (Accepted) 
>  10:49:26.933   pjsua_core.c RX 571 bytes Request msg NOTIFY/cseq=102 (rdata0x81738dc) from UDP 192.168.1.10:5060:
> NOTIFY sip:131 at 192.168.1.213:50060;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK03d9d68f;rport
> From: sip:131 at 192.168.1.10;tag=as2d7b4585
> To: sip:131 at localhost;tag=25e0000467458b6bc623
> Contact: 
> Call-ID: 25e0000567458b6bc623
> CSeq: 102 NOTIFY
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Event: refer;id=1714636917
> Subscription-state: active
> Content-Type: message/sipfrag;version=2.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 21
> 
> SIP/2.0 183 Ringing
> 
> --end msg--
>  10:49:26.933   pjsua_core.c TX 447 bytes Response msg 200/NOTIFY/cseq=102 (tdta0x817bafc) to UDP 192.168.1.10:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.1.10:5060;rport=5060;received=192.168.1.10;branch=z9hG4bK03d9d68f
> Call-ID: 25e0000567458b6bc623
> From: ;tag=as2d7b4585
> To: ;tag=25e0000467458b6bc623
> CSeq: 102 NOTIFY
> Contact: 
> Allow: INVITE, ACK, BYE, CANCEL, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS
> Supported: replaces, norefersub
> Content-Length:  0
> 
> 
> --end msg--
>  10:49:26.933 evsub0x817da74 Subscription state changed ACCEPTED --> ACTIVE
>  10:49:26.933       makeCall Call 0: transfer status=183 ( Ringing) 
>  10:49:26.933   pjsua_core.c RX 588 bytes Request msg NOTIFY/cseq=103 (rdata0x81738dc) from UDP 192.168.1.10:5060:
> NOTIFY sip:131 at 192.168.1.213:50060;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1c6b1e6e;rport
> From: sip:131 at 192.168.1.10;tag=as2d7b4585
> To: sip:131 at localhost;tag=25e0000467458b6bc623
> Contact: 
> Call-ID: 25e0000567458b6bc623
> CSeq: 103 NOTIFY
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Event: refer;id=1714636917
> Subscription-state: terminated;reason=noresource
> Content-Type: message/sipfrag;version=2.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 16
> 
> SIP/2.0 200 Ok
> 
> --end msg--
>  10:49:26.933   pjsua_core.c TX 447 bytes Response msg 200/NOTIFY/cseq=103 (tdta0x8181e3c) to UDP 192.168.1.10:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.1.10:5060;rport=5060;received=192.168.1.10;branch=z9hG4bK1c6b1e6e
> Call-ID: 25e0000567458b6bc623
> From: ;tag=as2d7b4585
> To: ;tag=25e0000467458b6bc623
> CSeq: 103 NOTIFY
> Contact: 
> Allow: INVITE, ACK, BYE, CANCEL, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS
> Supported: replaces, norefersub
> Content-Length:  0
> 
> 
> --end msg--
>  10:49:26.933 evsub0x817da74 Subscription state changed ACTIVE --> TERMINATED
>  10:49:26.933   pjsua_call.c Xfer client subscription terminated
>  10:49:26.933       makeCall Call 0: transfer status=200 ( Ok) [final]
>  10:49:26.933       makeCall Call 0: call transfered successfully, disconnecting call
>  10:49:26.934   pjsua_core.c TX 299 bytes Request msg BYE/cseq=1714636918 (tdta0x8183dec) to UDP 192.168.1.10:5060:
> BYE sip:131 at 192.168.1.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000a67458b6bc623
> Max-Forwards: 70
> From: sip:131 at localhost;tag=25e0000467458b6bc623
> To: sip:131 at 192.168.1.10;tag=as2d7b4585
> Call-ID: 25e0000567458b6bc623
> CSeq: 1714636918 BYE
> Content-Length:  0
> 
> 
> --end msg--
>  10:49:26.934     sip_regc.c Unable to send request, regc has another transaction pending
>  10:49:26.934    pjsua_acc.c Unable to create/send REGISTER: Object is busy (PJSIP_EBUSY) [status=171001]
>  10:49:26.934    pjsua_acc.c Account id 0 deleted
>  10:49:26.940   pjsua_core.c RX 439 bytes Response msg 200/BYE/cseq=1714636918 (rdata0x81738dc) from UDP 192.168.1.10:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.1.213:50060;branch=z9hG4bKPj25e0000a67458b6bc623;received=192.168.1.213;rport=50060
> From: sip:131 at localhost;tag=25e0000467458b6bc623
> To: sip:131 at 192.168.1.10;tag=as2d7b4585
> Call-ID: 25e0000567458b6bc623
> CSeq: 1714636918 BYE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: 
> Content-Length: 0






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