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VoIPGRID / VialerSIPLib

An Objective-c wrapper for PJSIP
GNU General Public License v3.0
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Audio cut-out problem after 3.3.0 #170

Closed ismaiI1 closed 5 years ago

ismaiI1 commented 5 years ago

I think, After 3.3.0, this library has audio cut-out issue.. when we make call, it's sending re-invite every 2 second.. Media is update so Call mute not working (#165) I will use 3.2.0. I hope this problem will be solved in next versions.. Thanks..

3.2.0 is working perfectly.

Version

3.4.2

Stacktrace / Error message

  💚 DEBUG   13:45:46.079 [VSLEndpoint void logCallBack(int, const char *, int):536] pjsua_media.c  .....Call 0: updating media..
  💚 DEBUG   13:45:46.079 [VSLEndpoint void logCallBack(int, const char *, int):536]          pjsua_media.c  .......Media stream call00:0 is destroyed
  💚 DEBUG   13:45:46.079 [VSLEndpoint void logCallBack(int, const char *, int):536]            pjsua_aud.c  ......Audio channel update..
  💚 DEBUG   13:45:46.080 [VSLEndpoint void logCallBack(int, const char *, int):536]        strm0x10c8ca028  .......Encoder stream started
  💚 DEBUG   13:45:46.080 [VSLEndpoint void logCallBack(int, const char *, int):536]        strm0x10c8ca028  .......Decoder stream started
  💚 DEBUG   13:45:46.080 [VSLEndpoint void logCallBack(int, const char *, int):536]          pjsua_media.c  ......Audio updated, stream #0: PCMA (sendrecv)
  💜 VERBOSE 13:45:46.080 [VSLEndpoint void onCallMediaState(pjsua_call_id):593] Received MediaState update for call:15DC2D8C-F31E-49B2-8372-24A64BCC47A6
  💜 VERBOSE 13:45:46.080 [VSLCall -[VSLCall mediaStateChanged:]:406] Media State Changed from VSLMediaStateActive to VSLMediaStateActive
  💙 INFO    13:45:46.080 [VSLRingback -[VSLRingback stop]:94] Stop ringback, isPlaying: NO
  💚 DEBUG   13:45:46.080 [VSLEndpoint void logCallBack(int, const char *, int):536]            pjsua_aud.c  .....Conf connect: 2 --> 0
  💚 DEBUG   13:45:46.080 [VSLEndpoint void logCallBack(int, const char *, int):536]           conference.c  ......Port 2 (sip:XXXXXXXXXX@sip.******.com) transmitting to port 0 (iPhone IO device)
  💚 DEBUG   13:45:46.081 [VSLEndpoint void logCallBack(int, const char *, int):536]            pjsua_aud.c  .....Conf connect: 0 --> 2
  💚 DEBUG   13:45:46.081 [VSLEndpoint void logCallBack(int, const char *, int):536]           conference.c  ......Port 0 (iPhone IO device) transmitting to port 2 (sip:XXXXXXXXXX@sip.************)
  💚 DEBUG   13:45:46.081 [VSLEndpoint void logCallBack(int, const char *, int):536]           pjsua_core.c  .....TX 626 bytes Request msg ACK/cseq=6343 (tdta0x10caf54a8) to UDP XXX.XXX.XXX.189:5060:
  ACK sip:100035XXXXXXXXXX@XXX.XXX.XXX.202:5060 SIP/2.0
  Via: SIP/2.0/UDP 192.168.0.66:40530;rport;branch=z9hG4bKPj9Yz2x1eyorup87tKbme-QKqr7WWmTuI8
  Max-Forwards: 70
  From: sip:XXXYYYZZZ@sip.***********;tag=yxznlh-8SF2x4TiQif5ULDfmnK4TLWpH
  To: sip:XXXXXXXXXX@sip.************;tag=as26a4f05d
  Call-ID: DTYEq-C16HB4DYMP4br8B83fKQozV9EY
  CSeq: 6343 ACK
  Route: <sip:XXX.XXX.XXX.189;lr;ftag=yxznlh-8SF2x4TiQif5ULDfmnK4TLWpH>
  Route: <sip:XXX.XXX.XXX.102:5061;lr;ftag=yxznlh-8SF2x4TiQif5ULDfmnK4TLWpH;did=07b.eea1;vsf=AAAAAAEFCAYAAAAFBgB2RSlBFltLTF9dWgBSX18udHI-;vst=AAAAAAUBBwEKAQ4GBARxcUtcXhZWS0NJQl0cY29tLnRy>
  Content-Length:  0

  --end msg--
  💚 DEBUG   13:45:46.088 [VSLEndpoint void logCallBack(int, const char *, int):536]           pjsua_call.c  .Call 0 sending re-INVITE for updating media session to use only one codec
  💚 DEBUG   13:45:46.089 [VSLEndpoint void logCallBack(int, const char *, int):536]           pjsua_core.c  ....TX 1223 bytes Request msg INVITE/cseq=6344 (tdta0x10cb684a8) to UDP XXX.XXX.XXX.189:5060:
  INVITE sip:100035XXXXXXXXXX@XXX.XXX.XXX.202:5060 SIP/2.0
  Via: SIP/2.0/UDP 192.168.0.66:40530;rport;branch=z9hG4bKPjfZ9bySWW9x82aF8EkhRABK7sbTQUp6vb 
  Max-Forwards: 70
  From: sip:XXXYYYZZZ@sip.***********;tag=yxznlh-8SF2x4TiQif5ULDfmnK4TLWpH
  To: sip:XXXXXXXXXX@sip.************;tag=as26a4f05d
  Contact: <sip:XXXYYYZZZ@192.168.0.66:40530;ob>
  Call-ID: DTYEq-C16HB4DYMP4br8B83fKQozV9EY
  CSeq: 6344 INVITE
  Route: <sip:XXX.XXX.XXX.189;lr;ftag=yxznlh-8SF2x4TiQif5ULDfmnK4TLWpH>
  Route: <sip:XXX.XXX.XXX.102:5061;lr;ftag=yxznlh-8SF2x4TiQif5ULDfmnK4TLWpH;did=07b.eea1;vsf=AAAAAAEFCAYAAAAFBgB2RSlBFltLTF9dWgBSX18udHI-;vst=AAAAAAUBBwEKAQ4GBARxcUtcXhZWS0NJQl0cY29tLnRy>

  Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  Supported: replaces, 100rel, timer, norefersub
  Session-Expires: 1800;refresher=uas
  Min-SE: 90
  Content-Type: application/sdp
  Content-Length:   313

  v=0
  o=- 3754377934 3754377940 IN IP4 192.168.2.42
  s=pjmedia
  b=AS:84
  t=0 0
  a=X-nat:0
  m=audio 4004 RTP/AVP 8 96
  c=IN IP4 192.168.2.42
  b=TIAS:64000
  a=rtcp:4005 IN IP4 192.168.2.42
  a=ssrc:1278702608 cname:567fd8ea637a1703
  a=rtpmap:8 PCMA/8000
  a=rtpmap:96 telephone-event/8000
  a=fmtp:96 0-16
  a=sendrecv

  --end msg--
  💚 DEBUG   13:45:46.114 [VSLEndpoint void logCallBack(int, const char *, int):536]           pjsua_core.c  .RX 402 bytes Response msg 100/INVITE/cseq=6344 (rdata0x10ccd5828) from UDP XXX.XXX.XXX.189:5060:
  SIP/2.0 100 trying -- your call is important to us

  Via: SIP/2.0/UDP 192.168.0.66:40530;rport=40530;branch=z9hG4bKPjfZ9bySWW9x82aF8EkhRABK7sbTQUp6vb;received=192.168.0.66
  From: sip:XXXYYYZZZ@sip.***********;tag=yxznlh-8SF2x4TiQif5ULDfmnK4TLWpH
  To: sip:XXXXXXXXXX@sip.************;tag=as26a4f05d
  Call-ID: DTYEq-C16HB4DYMP4br8B83fKQozV9EY
  CSeq: 6344 INVITE
  Server: NetGSM
  Content-Length: 0

  --end msg--
  💚 DEBUG   13:45:46.119 [VSLEndpoint void logCallBack(int, const char *, int):536]           pjsua_core.c  .RX 854 bytes Response msg 200/INVITE/cseq=6344 (rdata0x10ccd5828) from UDP XXX.XXX.XXX.189:5060:
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.0.66:40530;received=192.168.0.66;rport=40530;branch=z9hG4bKPjfZ9bySWW9x82aF8EkhRABK7sbTQUp6vb
  From: sip:XXXYYYZZZ@sip.***********;tag=yxznlh-8SF2x4TiQif5ULDfmnK4TLWpH
  To: sip:XXXXXXXXXX@sip.************;tag=as26a4f05d
  Call-ID: DTYEq-C16HB4DYMP4br8B83fKQozV9EY
  CSeq: 6344 INVITE
  Server: sipgw2
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  Supported: replaces, timer
  Session-Expires: 1800;refresher=uas
  Contact: <sip:100035XXXXXXXXXX@XXX.XXX.XXX.202:5060>
  Content-Type: application/sdp
  Require: timer
  Content-Length: 231

  v=0
  o=root 576436610 576436615 IN IP4 XXX.XXX.XXX.202
  s=Asterisk PBX 11.25.3
  c=IN IP4 XXX.XXX.XXX.202
  t=0 0
  m=audio 15122 RTP/AVP 8 96
  a=rtpmap:8 PCMA/8000
  a=rtpmap:96 telephone-event/8000
  a=fmtp:96 0-16
  a=ptime:20
  a=sendrecv
jeremynorman89 commented 5 years ago

I'm closing this as it's referring to the same underlying issues as #165