search

sssun / webrtc2sip

Automatically exported from code.google.com/p/webrtc2sip
0 stars 0 forks source link

SIP Reinvite not working when using sipml5 and mobicents #117

Open GoogleCodeExporter opened 8 years ago

GoogleCodeExporter commented 8 years ago 
What steps will reproduce the problem?

1. Place a sipml5 call to Cisco Telepresence Server via webrtc2sip and mobicents
2. Cisco TelePresence Server answer call and then send a SIP Reinvite with SDP.
3. webrtc2sip does not forward SIP reinvite back to sipml5.

Reference
https://groups.google.com/forum/#!topic/doubango/eCFu5Nw3lLY

What is the expected output? What do you see instead?

webrtc2sip forwarding SIP reinvite to sipml5 client

What version of the product are you using? On what operating system?

Please provide server logs with DEBUG level equal to INFO

Please provide browser logs

Original issue reported on code.google.com by gascagon...@gmail.com on 8 Aug 2013 at 5:37

GoogleCodeExporter commented 8 years ago 
Error I get:
*INFO: Cannot find peer with remote IP/Port=0.0.0.0/5060, connecting to the 
destination...
**WARN: function: "tnet_sockfd_connectto()" 
file: "src/tnet_utils.c" 
line: "1612" 
MSG: 
TNET_ERROR_WOULDBLOCK/TNET_ERROR_ISCONN/TNET_ERROR_INPROGRESS/TNET_ERROR_EAGAIN 
 ==> use tnet_sockfd_waitUntilWritable.
*INFO: ioctlt(4), len=0 returned zero or failed
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- FD_ACCEPT(fd=33)
*INFO: Socket added[SIP transport]: fd=33, tail.count=5
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- TNET_POLLOUT
*INFO: Socket added[SIP transport]: fd=32, tail.count=6
*INFO: Socket added (external call) 32
*INFO: Add call-id = '10746737-3205-353f-2116-51413b75ca18' to peer with local 
fd = 32
*INFO: Data send requested but peer not connected yet...saving data
*INFO: State machine: Exec function failed. Moving to terminal state.
*INFO: === IST terminated ===
*INFO: *** IST destroyed ***
*INFO: Stream Peer accepted/connected - 33
*INFO: Stream Peer accepted/connected - 33
*INFO: PipeR event = 1
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- TNET_POLLOUT
*INFO: Stream Peer accepted/connected - 32
*INFO:

Original comment by gascagon...@gmail.com on 8 Aug 2013 at 5:39

GoogleCodeExporter commented 8 years ago 
When breaker is enabled webrtc2sip drops any reInvite, whether coming from any 
direction, due to which features like hold-resume, call transfer don't work 
properly. Please look into this issue

Original comment by amit.bha...@gmail.com on 8 Aug 2013 at 10:40

GoogleCodeExporter commented 8 years ago 
Can anyone solve this issue? because without the web breaker the audio don't 
flow between sip terminals. And can't transfer or put on hold the call is a 
real problem when trying to implement a websipphone...

Original comment by laar78...@gmail.com on 23 Apr 2014 at 5:20