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Applied several hacks to make sipml5 more stable #280

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@roginvs roginvs commented Jan 16, 2017

  • ICE Gathering timeout hack: This solves a problem when calls can delay up to 10 seconds. It happens when some of stun requests are dropped, for example if they are sent from interface without internet and etc.
  • Failed to set local answer sdp: In some cases sipml5 can cause this error in Chrome. Workaround is taken from https://code.google.com/archive/p/sipml5/issues/209 with a simple change to make hold/resume working again.
  • Fix for CallerId Name problems from Caller ID Name Not Present on Incoming Call to sipML5 #163
  • Indication for "Call completed elsewhere". Just simple lookup for such reason. This is useful in order to not to show such calls as missed

@ackvf
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ackvf commented Feb 9, 2017

Will you also fix the issue with new chrome?
https://medium.com/@nimbleape/webrtc-asterisk-and-chrome-57-a706fde33780#.79lke0ggg

@roginvs
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roginvs commented Feb 10, 2017

@vferko , thanks, Chrome 57 could be a surprise for us. Fixed in 4e4b728

@DoubangoTelecom
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4c97518

@alepolidori
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Hi @roginvs.
I've tried your patch with "Version 57.0.2987.37 beta (64-bit)" on linux, but I have the problem to call: it fails.

I've tried the code from your repo https://github.com/roginvs/sipml5/commits/patches-to-origin and from the doubango master https://github.com/DoubangoTelecom/sipml5/commits/master.

I'm not sure what I've done wrong.

Scenario:
extension 351 and 352 registered and I make a call from 351 to 352.
PBX: asterisk
During the INVITE the "a=rtcp-mux" is present.

The cause seems to be the busy status:

...
SIPml-api.js:1 recv=SIP/2.0 486 Busy Here
Via: SIP/2.0/WSS df7asdf3ls0d.invalid;rport=35256;received=<IP>;branch=z9hG4bKufOTsEaN55RBGWg2H8zgwkQWWNrHOxyt
From: "351"<sip:351@<SERVER>>;tag=i0fcsdfsEsTzF32ehgAF
To: <sip:352@<SERVER>>;tag=as3916794b
Call-ID: 284ae60a-7c29-0e49-afae-946ed3d94948
CSeq: 9742 INVITE
Content-Length: 0
Server: NEWBRAND2.10.95(11.22.0)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
...

Do you need all the debug messages ?

If I make a call to an external mobile phone, the call is ok, but when I answer no audio is present.

(With current Chrome 56 stable all work correctly.)

Could you give me some tips ?
Thank you

@roginvs
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roginvs commented Feb 13, 2017

@alepolidori , I have same Chrome version [57.0.2987.37 beta (64-bit)], but on Windows 10, and it works.
Can you show Chrome logs? Also maybe your browser loaded cached version of old library.
You can also try demo from github pages

@alepolidori
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alepolidori commented Feb 13, 2017

@roginvs
I've tried also in Windows 10, but it fails.
When I call from chrome stable to chrome beta the call fails immediately.
If I call from chrome beta to chrome stable the call will connect but no audio is present.

Following is the log of call using both chrome beta
REGISTER phase:

SIPML5 API version = 2.1.3
SIPml-api-fixed.js:1 User-Agent=Mozilla/5.0 (X11; Linux x86_64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/57.0.2987.37 Safari/537.36
SIPml-api-fixed.js:1 WebSocket supported = yes
SIPml-api-fixed.js:1 Navigator friendly name = chrome
SIPml-api-fixed.js:1 OS friendly name = linux
SIPml-api-fixed.js:1 Have WebRTC = yes
SIPml-api-fixed.js:1 Have GUM = yes
SIPml-api-fixed.js:1 Engine initialized
xt_interface.js:110 connected!
SIPml-api-fixed.js:1 s_websocket_server_url=wss://192.168.5.238:8089/ws
SIPml-api-fixed.js:1 s_sip_outboundproxy_url=udp://192.168.5.238:5060
SIPml-api-fixed.js:1 b_rtcweb_breaker_enabled=no
SIPml-api-fixed.js:1 b_click2call_enabled=no
SIPml-api-fixed.js:1 b_early_ims=yes
SIPml-api-fixed.js:1 b_enable_media_stream_cache=no
SIPml-api-fixed.js:1 o_bandwidth={}
SIPml-api-fixed.js:1 o_video_size={}
SIPml-api-fixed.js:1 SIP stack start: proxy='ns313841.ovh.net:10062', realm='<sip:192.168.5.238>', impi='211', impu='"211"<sip:[email protected]>'
SIPml-api-fixed.js:1 Connecting to 'wss://192.168.5.238:8089/ws'
SIPml-api-fixed.js:1 __tsip_transport_ws_onopen
SIPml-api-fixed.js:1 State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister
SIPml-api-fixed.js:1 SEND: REGISTER sip:192.168.5.238 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKK2QzQ2g3iL6oOFOCHmrL1dc1fU4b5Qow;rport
From: "211"<sip:[email protected]>;tag=UiqeAOhEuzFzXC5Fvmqx
To: "211"<sip:[email protected]>
Contact: "211"<sips:[email protected];rtcweb-breaker=no;transport=wss>;expires=1800;click2call=no
Call-ID: c7f345c2-5e4a-10ca-b242-7f12191042e3
CSeq: 36072 REGISTER
Content-Length: 0
Route: <sip:192.168.5.238:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: SIP Phone
Supported: path


SIPml-api-fixed.js:1 __tsip_transport_ws_onmessage
SIPml-api-fixed.js:1 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=54006;received=192.168.5.60;branch=z9hG4bKK2QzQ2g3iL6oOFOCHmrL1dc1fU4b5Qow
From: "211"<sip:[email protected]>;tag=UiqeAOhEuzFzXC5Fvmqx
To: "211"<sip:[email protected]>;tag=as5579d82b
Call-ID: c7f345c2-5e4a-10ca-b242-7f12191042e3
CSeq: 36072 REGISTER
Content-Length: 0
Server: NEWBRAND2.10.95(11.22.0)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="asterisk",nonce="3b025ef8",stale=FALSE,algorithm=MD5


SIPml-api-fixed.js:1 State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
SIPml-api-fixed.js:1 SEND: REGISTER sip:192.168.5.238 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKaaQcVIo3auNyCCcmw6EwPezEtl58VhIn;rport
From: "211"<sip:[email protected]>;tag=UiqeAOhEuzFzXC5Fvmqx
To: "211"<sip:[email protected]>
Contact: "211"<sips:[email protected];rtcweb-breaker=no;transport=wss>;expires=1800;click2call=no
Call-ID: c7f345c2-5e4a-10ca-b242-7f12191042e3
CSeq: 36073 REGISTER
Content-Length: 0
Route: <sip:192.168.5.238:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="211",realm="asterisk",nonce="3b025ef8",uri="sip:192.168.5.238",response="2c00be7aa4faf848e5f20fff264025f0",algorithm=MD5
User-Agent: SIP Phone
Supported: path


SIPml-api-fixed.js:1 __tsip_transport_ws_onmessage
SIPml-api-fixed.js:1 recv=OPTIONS sips:[email protected];rtcweb-breaker=no;transport=wss SIP/2.0
Via: SIP/2.0/WS 192.168.5.238:5060;rport;branch=z9hG4bK5e8c776b
From: "Unknown"<sip:[email protected]>;tag=as2e5cd6b4
To: <sips:[email protected];rtcweb-breaker=no;transport=wss>
Contact: <sip:[email protected]:5060;transport=WS>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0
Max-Forwards: 70
User-Agent: NEWBRAND2.10.95(11.22.0)
Date: 30 Jan 2017 18:55:26 GMT;30
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer


SIPml-api-fixed.js:1 SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WS 192.168.5.238:5060;rport=5060;branch=z9hG4bK5e8c776b
From: "Unknown"<sip:[email protected]>;tag=as2e5cd6b4
To: <sips:[email protected];rtcweb-breaker=no;transport=wss>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0


SIPml-api-fixed.js:1 __tsip_transport_ws_onmessage
SIPml-api-fixed.js:1 recv=SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=54006;received=192.168.5.60;branch=z9hG4bKaaQcVIo3auNyCCcmw6EwPezEtl58VhIn
From: "211"<sip:[email protected]>;tag=UiqeAOhEuzFzXC5Fvmqx
To: "211"<sip:[email protected]>;tag=as5579d82b
Contact: <sips:[email protected];rtcweb-breaker=no;transport=wss>;expires=1800
Call-ID: c7f345c2-5e4a-10ca-b242-7f12191042e3
CSeq: 36073 REGISTER
Expires: 1800
Content-Length: 0
Server: NEWBRAND2.10.95(11.22.0)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 30 Jan 2017 18:55:26 GMT;30


SIPml-api-fixed.js:1 State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx
SIPml-api-fixed.js:1 __tsip_transport_ws_onmessage
SIPml-api-fixed.js:1 recv=NOTIFY sips:[email protected];rtcweb-breaker=no;transport=wss SIP/2.0
Via: SIP/2.0/WS 192.168.5.238:5060;rport;branch=z9hG4bK73d04998
From: "Unknown"<sip:[email protected]>;tag=as122e62b1
To: <sips:[email protected];rtcweb-breaker=no;transport=wss>
Contact: <sip:[email protected]:5060;transport=WS>
Call-ID: [email protected]:5060
CSeq: 102 NOTIFY
Content-Type: application/simple-message-summary
Content-Length: 101
Max-Forwards: 70
User-Agent: NEWBRAND2.10.95(11.22.0)
Event: message-summary

Messages-Waiting: no
Message-Account: sip:*[email protected];transport=WS
Voice-Message: 0/0 (0/0)

SIPml-api-fixed.js:1 SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/WS 192.168.5.238:5060;rport=5060;branch=z9hG4bK73d04998
From: "Unknown"<sip:[email protected]>;tag=as122e62b1
To: <sips:[email protected];rtcweb-breaker=no;transport=wss>
Call-ID: [email protected]:5060
CSeq: 102 NOTIFY
Content-Length: 0

CALL phase:

State machine: c0000_Started_2_Outgoing_X_oINVITE
SIPml-api-fixed.js:1 ICE servers:[]
SIPml-api-fixed.js:1 onGetUserMediaSuccess
SIPml-api-fixed.js:1 createOffer
SIPml-api-fixed.js:1 onCreateSdpSuccess
SIPml-api-fixed.js:1 onNegotiationNeeded
SIPml-api-fixed.js:1 onSetLocalDescriptionSuccess
SIPml-api-fixed.js:1 onSignalingstateChange:have-local-offer
2SIPml-api-fixed.js:1 onIceCandidate = gathering
SIPml-api-fixed.js:1 onIceCandidate = complete
SIPml-api-fixed.js:1 ICE GATHERING COMPLETED!
SIPml-api-fixed.js:1 onIceGatheringCompleted
SIPml-api-fixed.js:1 SEND: INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKuXRl5UtLxTIwZeQc8R7wxoKPCPHCPewQ;rport
From: "211"<sip:[email protected]>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:[email protected]>
Contact: "211"<sips:[email protected];rtcweb-breaker=no;click2call=no;transport=wss>
Call-ID: 51bb4209-df2a-0d31-f371-ee10436b1fc3
CSeq: 38709 INVITE
Content-Type: application/sdp
Content-Length: 1462
Route: <sip:192.168.5.238:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: SIP Phone

v=0
o=- 4576999717784606000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS LHDDKMg1GtmJLMkzIybxiOsCSEvs6DHZskWu
m=audio 54967 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 192.168.5.60
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3043313387 1 udp 2122260223 192.168.5.60 54967 typ host generation 0 network-id 1
a=candidate:4226202139 1 tcp 1518280447 192.168.5.60 9 typ host tcptype active generation 0 network-id 1
a=ice-ufrag:eICW
a=ice-pwd:0cVFayVWJr174z7QMArUkVG7
a=fingerprint:sha-256 88:9D:DC:AC:76:22:4C:8D:74:3E:42:39:D5:24:42:01:4C:CD:57:CE:5F:AD:9B:33:42:AE:10:8A:FB:91:A2:08
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1860465271 cname:Ci2WNzSeFbC5p86g
a=ssrc:1860465271 msid:LHDDKMg1GtmJLMkzIybxiOsCSEvs6DHZskWu a55dab6c-fb39-4b7b-ac61-c071ac9d1392
a=ssrc:1860465271 mslabel:LHDDKMg1GtmJLMkzIybxiOsCSEvs6DHZskWu
a=ssrc:1860465271 label:a55dab6c-fb39-4b7b-ac61-c071ac9d1392

SIPml-api-fixed.js:1 __tsip_transport_ws_onmessage
SIPml-api-fixed.js:1 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=54006;received=192.168.5.60;branch=z9hG4bKuXRl5UtLxTIwZeQc8R7wxoKPCPHCPewQ
From: "211"<sip:[email protected]>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:[email protected]>;tag=as5262ba5f
Call-ID: 51bb4209-df2a-0d31-f371-ee10436b1fc3
CSeq: 38709 INVITE
Content-Length: 0
Server: NEWBRAND2.10.95(11.22.0)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="asterisk",nonce="2accdeb5",stale=FALSE,algorithm=MD5


SIPml-api-fixed.js:1 SEND: ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKuXRl5UtLxTIwZeQc8R7wxoKPCPHCPewQ;rport
From: "211"<sip:[email protected]>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:[email protected]>;tag=as5262ba5f
Call-ID: 51bb4209-df2a-0d31-f371-ee10436b1fc3
CSeq: 38709 ACK
Content-Length: 0
Route: <sip:192.168.5.238:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70


SIPml-api-fixed.js:1 State machine: x0000_Any_2_Any_X_i401_407_INVITE
SIPml-api-fixed.js:1 SEND: INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKrIUG9WrD18wBlkibSO8xHcFlPZGCig7K;rport
From: "211"<sip:[email protected]>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:[email protected]>
Contact: "211"<sips:[email protected];rtcweb-breaker=no;click2call=no;transport=wss>
Call-ID: 51bb4209-df2a-0d31-f371-ee10436b1fc3
CSeq: 38710 INVITE
Content-Type: application/sdp
Content-Length: 1462
Route: <sip:192.168.5.238:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="211",realm="asterisk",nonce="2accdeb5",uri="sip:[email protected]",response="43ad0e7a4594bca72fbee52fa248c807",algorithm=MD5
User-Agent: SIP Phone

v=0
o=- 4576999717784606000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS LHDDKMg1GtmJLMkzIybxiOsCSEvs6DHZskWu
m=audio 54967 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 192.168.5.60
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3043313387 1 udp 2122260223 192.168.5.60 54967 typ host generation 0 network-id 1
a=candidate:4226202139 1 tcp 1518280447 192.168.5.60 9 typ host tcptype active generation 0 network-id 1
a=ice-ufrag:eICW
a=ice-pwd:0cVFayVWJr174z7QMArUkVG7
a=fingerprint:sha-256 88:9D:DC:AC:76:22:4C:8D:74:3E:42:39:D5:24:42:01:4C:CD:57:CE:5F:AD:9B:33:42:AE:10:8A:FB:91:A2:08
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1860465271 cname:Ci2WNzSeFbC5p86g
a=ssrc:1860465271 msid:LHDDKMg1GtmJLMkzIybxiOsCSEvs6DHZskWu a55dab6c-fb39-4b7b-ac61-c071ac9d1392
a=ssrc:1860465271 mslabel:LHDDKMg1GtmJLMkzIybxiOsCSEvs6DHZskWu
a=ssrc:1860465271 label:a55dab6c-fb39-4b7b-ac61-c071ac9d1392

SIPml-api-fixed.js:1 __tsip_transport_ws_onmessage
SIPml-api-fixed.js:1 recv=SIP/2.0 100 Trying
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=54006;received=192.168.5.60;branch=z9hG4bKrIUG9WrD18wBlkibSO8xHcFlPZGCig7K
From: "211"<sip:[email protected]>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060;transport=WS>
Call-ID: 51bb4209-df2a-0d31-f371-ee10436b1fc3
CSeq: 38710 INVITE
Content-Length: 0
Server: NEWBRAND2.10.95(11.22.0)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer


SIPml-api-fixed.js:1 State machine: x0000_Any_2_Any_X_i1xx
SIPml-api-fixed.js:1 __tsip_transport_ws_onmessage
SIPml-api-fixed.js:1 recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=54006;received=192.168.5.60;branch=z9hG4bKrIUG9WrD18wBlkibSO8xHcFlPZGCig7K
From: "211"<sip:[email protected]>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:[email protected]>;tag=as0af9675b
Contact: <sip:[email protected]:5060;transport=WS>
Call-ID: 51bb4209-df2a-0d31-f371-ee10436b1fc3
CSeq: 38710 INVITE
Content-Length: 0
Server: NEWBRAND2.10.95(11.22.0)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
P-Asserted-Identity: "212"<sip:[email protected]>


SIPml-api-fixed.js:1 State machine: x0000_Any_2_Any_X_i1xx
SIPml-api-fixed.js:1 __tsip_transport_ws_onmessage
SIPml-api-fixed.js:1 recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=54006;received=192.168.5.60;branch=z9hG4bKrIUG9WrD18wBlkibSO8xHcFlPZGCig7K
From: "211"<sip:[email protected]>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:[email protected]>;tag=as0af9675b
Contact: <sip:[email protected]:5060;transport=WS>
Call-ID: 51bb4209-df2a-0d31-f371-ee10436b1fc3
CSeq: 38710 INVITE
Content-Length: 0
Server: NEWBRAND2.10.95(11.22.0)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer


SIPml-api-fixed.js:1 State machine: x0000_Any_2_Any_X_i1xx
SIPml-api-fixed.js:1 __tsip_transport_ws_onmessage
SIPml-api-fixed.js:1 recv=SIP/2.0 486 Busy Here
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=54006;received=192.168.5.60;branch=z9hG4bKrIUG9WrD18wBlkibSO8xHcFlPZGCig7K
From: "211"<sip:[email protected]>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:[email protected]>;tag=as0af9675b
Call-ID: 51bb4209-df2a-0d31-f371-ee10436b1fc3
CSeq: 38710 INVITE
Content-Length: 0
Server: NEWBRAND2.10.95(11.22.0)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21


SIPml-api-fixed.js:1 SEND: ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKrIUG9WrD18wBlkibSO8xHcFlPZGCig7K;rport
From: "211"<sip:[email protected]>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:[email protected]>;tag=as0af9675b
Call-ID: 51bb4209-df2a-0d31-f371-ee10436b1fc3
CSeq: 38710 ACK
Content-Length: 0
Route: <sip:192.168.5.238:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70


SIPml-api-fixed.js:1 State machine: c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE
SIPml-api-fixed.js:1 === INVITE Dialog terminated ===
SIPml-api-fixed.js:1 PeerConnection::stop()
SIPml-api-fixed.js:1 onIceGatheringCompleted

@roginvs
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roginvs commented Feb 13, 2017

And what about the opposite peer? Logs on this side does not shows anything :(
I just tried to call from Chrome 56 to Chrome 57 (both computers are Windows 10 x64), and everything were fine [opposite direction too]. I tested on asterisk 11.22.0 (with opus patch + other several patches).
I suppose "directmedia" option in asterisk can cause this, but this is just a theory.
After 3 hours I will be able to create a temp host with asterisk for you, will you try on it? Write to me in skype vasya_rogin if you want

@alepolidori
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I try to recap by providing more detailed info to avoid confusion.
I used asterisk 11.22.0 (without opus patch), Chrome 57 Beta under linux (but I've tried also win10)
The test is a simple call between two extension: from 211 to 212 (both webrtc sip phone).
I tested SIPml-api.js from repo "master" of "Doubango" repo and from "patches_to_origin" of "roginvs" repo.

Test 1 using "master" of "Doubango"
Test 2 using "patches_to_origin" of "roginvs"

As you can see the error seems to be the lack of "a=rtcp-mux" into the called extension.
I'm available for any type of test.

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4 participants