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DoubangoTelecom / sipml5

The world's first HTML5 SIP client (WebRTC)
BSD 3-Clause "New" or "Revised" License
944 stars 460 forks source link

Applied several hacks to make sipml5 more stable #280

Open roginvs opened 7 years ago

roginvs commented 7 years ago
ackvf commented 7 years ago

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

roginvs commented 7 years ago

@vferko , thanks, Chrome 57 could be a surprise for us. Fixed in 4e4b728c97df9e649affc0bcb3af75e6062de053

DoubangoTelecom commented 7 years ago

4c97518c4c8e54e82d28d484a7abe00b9d44d900

alepolidori commented 7 years ago

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 commented 7 years ago

@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 commented 7 years ago

@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:211@192.168.5.238>'
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:211@192.168.5.238>;tag=UiqeAOhEuzFzXC5Fvmqx
To: "211"<sip:211@192.168.5.238>
Contact: "211"<sips:211@df7jal23ls0d.invalid;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:211@192.168.5.238>;tag=UiqeAOhEuzFzXC5Fvmqx
To: "211"<sip:211@192.168.5.238>;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:211@192.168.5.238>;tag=UiqeAOhEuzFzXC5Fvmqx
To: "211"<sip:211@192.168.5.238>
Contact: "211"<sips:211@df7jal23ls0d.invalid;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:211@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss SIP/2.0
Via: SIP/2.0/WS 192.168.5.238:5060;rport;branch=z9hG4bK5e8c776b
From: "Unknown"<sip:Unknown@192.168.5.238>;tag=as2e5cd6b4
To: <sips:211@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>
Contact: <sip:Unknown@192.168.5.238:5060;transport=WS>
Call-ID: 6ef50c7d3a85cb154f506f703da08471@192.168.5.238:5060
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:Unknown@192.168.5.238>;tag=as2e5cd6b4
To: <sips:211@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>
Call-ID: 6ef50c7d3a85cb154f506f703da08471@192.168.5.238:5060
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:211@192.168.5.238>;tag=UiqeAOhEuzFzXC5Fvmqx
To: "211"<sip:211@192.168.5.238>;tag=as5579d82b
Contact: <sips:211@df7jal23ls0d.invalid;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:211@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss SIP/2.0
Via: SIP/2.0/WS 192.168.5.238:5060;rport;branch=z9hG4bK73d04998
From: "Unknown"<sip:Unknown@192.168.5.238>;tag=as122e62b1
To: <sips:211@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>
Contact: <sip:Unknown@192.168.5.238:5060;transport=WS>
Call-ID: 18c599e21da9ce2906ffe1931adb5c33@192.168.5.238:5060
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:*97@192.168.5.238;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:Unknown@192.168.5.238>;tag=as122e62b1
To: <sips:211@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>
Call-ID: 18c599e21da9ce2906ffe1931adb5c33@192.168.5.238:5060
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:212@192.168.5.238 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKuXRl5UtLxTIwZeQc8R7wxoKPCPHCPewQ;rport
From: "211"<sip:211@192.168.5.238>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:212@192.168.5.238>
Contact: "211"<sips:211@df7jal23ls0d.invalid;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:211@192.168.5.238>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:212@192.168.5.238>;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:212@192.168.5.238 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKuXRl5UtLxTIwZeQc8R7wxoKPCPHCPewQ;rport
From: "211"<sip:211@192.168.5.238>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:212@192.168.5.238>;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:212@192.168.5.238 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKrIUG9WrD18wBlkibSO8xHcFlPZGCig7K;rport
From: "211"<sip:211@192.168.5.238>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:212@192.168.5.238>
Contact: "211"<sips:211@df7jal23ls0d.invalid;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:212@192.168.5.238",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:211@192.168.5.238>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:212@192.168.5.238>
Contact: <sip:212@192.168.5.238:5060;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:211@192.168.5.238>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:212@192.168.5.238>;tag=as0af9675b
Contact: <sip:212@192.168.5.238:5060;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:212@192.168.5.238>

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:211@192.168.5.238>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:212@192.168.5.238>;tag=as0af9675b
Contact: <sip:212@192.168.5.238:5060;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:211@192.168.5.238>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:212@192.168.5.238>;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:212@192.168.5.238 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKrIUG9WrD18wBlkibSO8xHcFlPZGCig7K;rport
From: "211"<sip:211@192.168.5.238>;tag=cmBkTIeiTHC4iO4oNVZT
To: <sip:212@192.168.5.238>;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 commented 7 years ago

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 commented 7 years ago

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.