What steps will reproduce the problem?
1. Call is made from Our Proprietary Client to Chrome
2. Call is established Successfully. Audio and Video are streamed both sides.
What is the expected output? What do you see instead?
1. When BYE is sent from the browser, Call should end on our Client (Expected).
2. When BYE is sent from the browser, Audio and video streams are disconnected
but the
call does not end at our Client.(Observed)
a) When BYE is sent from the browser, WebRTC2Sip Gateway continuously makes DNS queries
to Junk Host Name and send BYE to IMS(According to webrtc2sip logs)
b) But according to Packet Traces, no BYE is forwarded by WebRTC2SIP Gateway to IMS.
CHROME WebRTC2SIP IMS OUR CLIENT
| | | |
| | | INVITE |
| | |<--------------|
| | INVITE | |
| |<--------------| |
| INVITE | | |
|<-------------| | |
| | | |
| 180 RINGING | | |
|------------->| | |
| | 180 RINGING | |
| |-------------->| |
| | | 180 RINGING |
| | |-------------->|
| 200 OK | | |
|------------->| | |
| | 200 OK | |
| |-------------->| |
| | | 200 OK |
| | |-------------->|
| | | |
| | | |
|--------------------------------------------->|
| RTP FLOW |
|<---------------------------------------------|
| | | |
| BYE | | |
|------------->| | |
| | | |
| 200 OK | | |
|<-------------| | |
| | | |
| | | |
| | | |
ISSUE:
When Call is ended from the Browser, BYE is sent from Chrome to WebRTC2Sip
Gateway but WebRTC2Sip Gateway does not forward BYE to IMS.
Observation:
1. Chrome Console Logs show BYE is sent to WebRTC2Sip Gateway.
2. WebRTC2Sip Gateway Logs (INFO Level) shows that BYE is received from Chrome.
3. WebRTC2Sip Gateway Logs (INFO Level) shows that BYE is being sent to Outbound
Proxy (a continuous DNS Query is made to a Junk Host Name before sending BYE,
also BYE is being sent multiple times). BUT Packet Traces show that NO BYE is
sent from WebRTC2Sip Gateway to Outbound Proxy(IMS).
4. IMS sends BYE to WebRTC2Sip Gateway after sometime due to the disconnection of RTP Flow (Timeout).
What version of the product are you using? On what operating system?
1. WebRTC Rev 100
2. Doubango Rev 985
3. Sipml5 Rev 197
Chrome(SIPML5) is running on Windows 7.
Webrtc2sip Gateway is running on RedHat6.
Please provide server logs with DEBUG level equal to INFO
1. webrtc2sip.zip
2. tcpDumpAtWebrtc2SIP.zip
Please provide browser logs
1. chromeConsole.zip
2. tcpDumpAtChrome.zip
1. Chrome Console Logs.
2. Packet Traces at Chrome.
3. WebRTC2Sip Logs (INFO Level).
4. Packet Traces at WebRTC2Sip Gateway.
Original issue reported on code.google.com by rohitsor...@gmail.com on 9 Aug 2013 at 7:14
Original issue reported on code.google.com by
rohitsor...@gmail.comon 9 Aug 2013 at 7:14Attachments: