letsgospinlab / webrtc2sip

Automatically exported from code.google.com/p/webrtc2sip
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BYE never reach the caller #56

Open GoogleCodeExporter opened 9 years ago

GoogleCodeExporter commented 9 years ago
Using Sipml5 and webrtc2sip gateway I am not able to terminate the call from 
the callee. Only the caller can terminate the call. 

Trace from the callee :

State machine: x0000_Any_2_Trying_X_oBYE tsk_utils.js:110
SEND: BYE 
sip:mobile@81.18.77.4:5080;rtcweb-breaker=no;transport=ws;ws-src-ip=188.25.108.2
06;ws-src-port=59533;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bK4GdXFO8DnvRGlUiW6uKgAsw3Ux1pN3RN;rport
From: <sip:bob@81.18.77.4>;tag=M3vf88lG7vIVXT9mNbRG
To: <sip:mobile@81.18.77.4>;tag=6wUL1mTAVHbd4PNZ2mMF
Call-ID: 17d9d1ea-b7d3-dd92-86cd-dd91ad2a16a0
CSeq: 53872 BYE
Content-Length: 0
Route: <sip:81.18.77.4:5060;lr;transport=udp>
Max-Forwards: 70
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;+sip.ice
Accept-Contact: *;language="en,fr"
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.01.14
Organization: Doubango Telecom

 tsk_utils.js:110
PeerConnection::stop() tsk_utils.js:110
==session event = terminating call.htm:720
__on_state_change tsk_utils.js:110
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
81.18.77.4:5080;rport=5080;branch=z9hG4bK4GdXFO8DnvRGlUiW6uKgAsw3Ux1pN3RN
From: <sip:bob@81.18.77.4>;tag=M3vf88lG7vIVXT9mNbRG
To: <sip:mobile@81.18.77.4>;tag=6wUL1mTAVHbd4PNZ2mMF
Call-ID: 17d9d1ea-b7d3-dd92-86cd-dd91ad2a16a0
CSeq: 53872 BYE
Content-Length: 0
Via: SIP/2.0/TCP 
188.25.108.206:59769;rport;branch=z9hG4bK4GdXFO8DnvRGlUiW6uKgAsw3Ux1pN3RN;ws-hac
ked=WS

BYE never reach at the caller.

if the caller hangs up, the following sequence happens:

SEND: BYE 
sip:bob@81.18.77.4:5080;transport=udp;ws-src-ip=188.25.108.206;ws-src-port=59769
;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKAJdTR6187h6RcfEef2aZPFF0rXUHhgEU;rport
From: <sip:mobile@81.18.77.4>;tag=PzRsqJBwmcAPhl3dcr1d
To: <sip:bob@81.18.77.4>;tag=gTMn72zo04ZFCzFQdmW4
Call-ID: 66ea53d2-5a11-e7a6-149b-d98a49eba6cb
CSeq: 23598 BYE
Content-Length: 0
Route: <sip:81.18.77.4:5060;lr;transport=udp>
Max-Forwards: 70
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;+sip.ice
Accept-Contact: *;language="en,fr"
Proxy-Authorization: Digest 
username="mobile",realm="81.18.77.4",nonce="13005863700:7550d915958e40c0e2efd7ce
c945d24a",uri="sip:bob@81.18.77.4:5080;transport=udp;ws-src-ip=188.25.108.206;ws
-src-port=59769;ws-src-proto=ws",response="9eba4a10ab935fbdba4c03ca23ec7a3a",alg
orithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.01.14
Organization: Doubango Telecom

 tsk_utils.js:110
PeerConnection::stop() tsk_utils.js:110
==session event = terminating call.htm:720
__on_state_change tsk_utils.js:110
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 200 OK
Via: SIP/2.0/UDP 
81.18.77.4:5080;rport=5080;branch=z9hG4bKAJdTR6187h6RcfEef2aZPFF0rXUHhgEU
From: <sip:mobile@81.18.77.4>;tag=PzRsqJBwmcAPhl3dcr1d
To: <sip:bob@81.18.77.4>;tag=gTMn72zo04ZFCzFQdmW4
Contact: 
<sip:bob@81.18.77.4:5080;transport=udp;ws-src-ip=188.25.108.206;ws-src-port=5976
9;ws-src-proto=ws>
Call-ID: 66ea53d2-5a11-e7a6-149b-d98a49eba6cb
CSeq: 23598 BYE
Content-Length: 0
Via: SIP/2.0/TCP 
188.25.108.206:59533;rport;branch=z9hG4bKAJdTR6187h6RcfEef2aZPFF0rXUHhgEU;ws-hac
ked=WS

 tsk_utils.js:110
State machine: x0000_Any_2_Terminated_X_i2xxBYE 

It looks like the mediaproxy doesn't set the right transport=udp for the 
caller. 

It works fine with resiprocate webrtc2sip. 

Need help

Thank you, 
Florin Popescu

Original issue reported on code.google.com by florin.p...@gmail.com on 20 Feb 2013 at 7:58

GoogleCodeExporter commented 9 years ago
Hi Florin,

have you managed to solve this issue? we're experiencing it as well, and it's 
driving us crazy.

Lorenzo

Original comment by lmini...@gmail.com on 8 Jul 2013 at 9:39

GoogleCodeExporter commented 9 years ago
Just to add a few more details about what we found out about the issue, it 
looks like a state machine issue. In our scenario, both the caller and the 
callee are handled by the webrtc2sip gateway, which is a frontend to an SBC. 
When the callee hangs up, webrtc2sip logs an "invite dialog terminated" event, 
and forwards the BYE to the SBC. When the BYE gets back to webrtc2sip to notify 
the caller, though, the gateway doesn't seem to be able to find the dialog 
associated to the Call-ID anymore ("No matching state machine found"), and so 
forwards the BYE as is to sipml5, which of course doesn't know how to handle it 
(wrong Call-ID) and ignores it. This makes me think that something in the 
application logic related to the first BYE (cleaning the session there) steps 
on the other session as well while it shouldn't, messing the state machine.

Anyone else met the same issue? It's been a while since this problem was first 
reported.

Lorenzo

Original comment by lmini...@gmail.com on 8 Jul 2013 at 9:54

GoogleCodeExporter commented 9 years ago

Original comment by boss...@yahoo.fr on 4 Aug 2013 at 1:16

GoogleCodeExporter commented 9 years ago
Fixed: https://code.google.com/p/doubango/issues/detail?id=292
Will be closed when fully tested and confirmed.

Original comment by boss...@yahoo.fr on 5 Aug 2013 at 5:03

GoogleCodeExporter commented 9 years ago
I also have the same problem, but when I use Websocket Secure. If I change the 
mode of transport to WS, it works correctly: Invoked event 'Call terminated' in 
the caller when the callee hangs.

Original comment by alvaro.r...@gmail.com on 13 Sep 2013 at 8:11

GoogleCodeExporter commented 9 years ago
Facing the same issue. Call won't terminate on the browser (caller) end when 
the callee terminates the call and I am using WSS. I am using WebRTC2SIP and 
have enabled RTC Web Breaker. The problem seems to disappear if I use WS 
instead of WSS. However, Firefox does not allow a call if WSS is used and the 
protocol of the site is not HTTPS. Can someone please share more details on 
this. Is this a known bug?

Original comment by johnsaj...@gmail.com on 19 Sep 2013 at 11:21

GoogleCodeExporter commented 9 years ago
I am also facing this issue.But only  when I use Websocket Secure.Unfortunately 
Firefox wont allow  ws for https connections.

Original comment by arununnikrishnan11@gmail.com on 20 Sep 2013 at 3:49

GoogleCodeExporter commented 9 years ago
hi.....
for Issue of WS in HTTPS ..In firefox type about:config...then search for 
websocket .Then make websocket for HTTPS true..It works for me..  

Original comment by soni8...@gmail.com on 1 Oct 2013 at 10:03

GoogleCodeExporter commented 9 years ago
[deleted comment]
GoogleCodeExporter commented 9 years ago
[deleted comment]
GoogleCodeExporter commented 9 years ago
[deleted comment]
GoogleCodeExporter commented 9 years ago
I have the same issue with ending call from callee when on wss and works ok 
with ws. (using chrome 39, asterisk 13). does anyone have solution? 

Original comment by matja...@gmail.com on 16 Jan 2015 at 1:39