Firefox nightly <-> Chrome

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Mamadou

unread,
Jan 14, 2013, 1:14:35 AM1/14/13
to discuss-webrtc
Hello,

For people using SIP as signaling protocol and a bit frustrated that
RTCWeb implementations from Mozilla and Google are not compatible yet,
we've updated our open source projects (sipml5.org and
webrtc2sip.org).
sipml5 allows Chome2Chrome and Firefox2Firefox calls without any
gateway. To make calls between Chrome and Firefox you will need the
enable the RTCWeb breaker from open source webrtc2sip gateway as
explained in the documentation.

Tested versions:
- Chrome stable "24.0.1312.52 m" on Windows Vista, 8 and OSX Lion
- Firefox Nightly 21.0a1 (2013-01-13) on Windows Vista, 8 and OSX
Lion. To enable WebRTC: http://code.google.com/p/sipml5/wiki/Enable_WebRTC
- If you're hosting your own webrtc2sip gateway, please note that it
requires SSL certificates (because of DTLS-SRTP).

Know working scenarios:
1. [chome]<--webrtc2sip-->[Firefox Nightly]
2. [chome]<--webrtc2sip-->[any SIP-legacy network]
3. [Firefox Nightly]<--webrtc2sip-->[any SIP-legacy network]
4. [chome]<->[chome]
5. [Firefox Nightly]<->[Firefox Nightly]

Know issues:
- The gateway fails to decode DTLS-SRTCP packets from Firefox Nightly.
As a direct consequence, the FIR and PLI requests are not honored.
We're working to fix this ASAP. More info at
http://code.google.com/p/doubango/issues/detail?id=194.

Happy testing :)

Regards,

Shachar

unread,
Jan 14, 2013, 3:00:36 AM1/14/13
to discuss...@googlegroups.com
Have you tested it with data channels?

Iwan Budi Kusnanto

unread,
Jan 14, 2013, 3:37:24 AM1/14/13
to discuss...@googlegroups.com
On Mon, Jan 14, 2013 at 1:14 PM, Mamadou <bos...@yahoo.fr> wrote:
> Hello,
>
> For people using SIP as signaling protocol and a bit frustrated that
> RTCWeb implementations from Mozilla and Google are not compatible yet,
> we've updated our open source projects (sipml5.org and
> webrtc2sip.org).
> sipml5 allows Chome2Chrome and Firefox2Firefox calls without any
> gateway. To make calls between Chrome and Firefox you will need the
> enable the RTCWeb breaker from open source webrtc2sip gateway as
> explained in the documentation.
>
> Tested versions:
> - Chrome stable "24.0.1312.52 m" on Windows Vista, 8 and OSX Lion
> - Firefox Nightly 21.0a1 (2013-01-13) on Windows Vista, 8 and OSX
> Lion. To enable WebRTC: http://code.google.com/p/sipml5/wiki/Enable_WebRTC
> - If you're hosting your own webrtc2sip gateway, please note that it
> requires SSL certificates (because of DTLS-SRTP).
>
> Know working scenarios:
> 1. [chome]<--webrtc2sip-->[Firefox Nightly]
> 2. [chome]<--webrtc2sip-->[any SIP-legacy network]


Hi Mamadou,
I use chrome 24 , calling asterisk 11.1.2, and no audio

tsk_utils.js:110
ICE GATHERING COMPLETED! tsk_utils.js:110
onIceGatheringCompleted tsk_utils.js:110
onIceGatheringCompleted but no local sdp request is pending

complete console log

YOUR ARE USING DEBUG CODE. PLEASE USE CODE UNDER 'release' FOLDER.
SIPml-api.js:19
User-Agent=Mozilla/5.0 (Macintosh; Intel Mac OS X 10_8_0)
AppleWebKit/537.17 (KHTML, like Gecko) Chrome/24.0.1312.52
Safari/537.17 tsk_utils.js:110
Navigator friendly name = chrome tsk_utils.js:110
OS friendly name = mac tsk_utils.js:110
s_websocket_server_url=(null) tsk_utils.js:110
s_sip_outboundproxy_url=(null) tsk_utils.js:110
b_rtcweb_breaker_enabled=yes tsk_utils.js:110
SIP stack start: proxy='sipml5.org:10062', realm='<sip:kuk.uk.to>',
impi='1061', impu='<sip:10...@kuk.uk.to>' tsk_utils.js:110
Connecting to 'wss://sipml5.org:10062' tsk_utils.js:110
==stack event = starting call.htm:632
__tsip_transport_ws_onopen tsk_utils.js:110
==stack event = started call.htm:632
State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister
tsk_utils.js:110
SEND: REGISTER sip:kuk.uk.to SIP/2.0
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bK6z6ZCcIsqg7rQXL7IcCqlRtge1eGw1E8;rport
From: <sip:10...@kuk.uk.to>;tag=UyU5RaiAzbxnnRBf4zTP
To: <sip:10...@kuk.uk.to>
Contact: "10...@kuk.uk.t"<sips:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;expires=200;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 92d82fd2-cb48-4e03-d1e6-2a614651cb09
CSeq: 32009 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.01.14
Organization: Doubango Telecom
Supported: path

tsk_utils.js:110
==session event = connecting call.htm:720
==session event = sent_request call.htm:720
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
87.106.69.240:10060;rport=10060;received=87.106.69.240;branch=z9hG4bK6z6ZCcIsqg7rQXL7IcCqlRtge1eGw1E8
From: <sip:10...@kuk.uk.to>;tag=UyU5RaiAzbxnnRBf4zTP
To: <sip:10...@kuk.uk.to>;tag=as008c7451
Call-ID: 92d82fd2-cb48-4e03-d1e6-2a614651cb09
CSeq: 32009 REGISTER
Content-Length: 0
Via: SIP/2.0/TCP
180.253.212.66:17589;rport;branch=z9hG4bK6z6ZCcIsqg7rQXL7IcCqlRtge1eGw1E8;ws-hacked=WSS
Server: Asterisk PBX 11.1.2
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest
realm="kuk.uk.to",nonce="62c6a6b4",stale=FALSE,algorithm=MD5

tsk_utils.js:110
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
tsk_utils.js:110
SEND: REGISTER sip:kuk.uk.to SIP/2.0
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bK5M5v4xgLelotJ5NDt8hwPcuOWEFrz5Xu;rport
From: <sip:10...@kuk.uk.to>;tag=UyU5RaiAzbxnnRBf4zTP
To: <sip:10...@kuk.uk.to>
Contact: "10...@kuk.uk.t"<sips:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;expires=200;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 92d82fd2-cb48-4e03-d1e6-2a614651cb09
CSeq: 32010 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest
username="1061",realm="kuk.uk.to",nonce="62c6a6b4",uri="sip:kuk.uk.to",response="2a8ff6db3255129083aa33df0cf039e9",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.01.14
Organization: Doubango Telecom
Supported: path

tsk_utils.js:110
==session event = sent_request call.htm:720
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 200 OK
Via: SIP/2.0/UDP
87.106.69.240:10060;rport=10060;received=87.106.69.240;branch=z9hG4bK5M5v4xgLelotJ5NDt8hwPcuOWEFrz5Xu
From: <sip:10...@kuk.uk.to>;tag=UyU5RaiAzbxnnRBf4zTP
To: <sip:10...@kuk.uk.to>;tag=as008c7451
Contact: <sip:10...@87.106.69.240:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.253.212.66;ws-src-port=17589;ws-src-proto=wss>;expires=200
Call-ID: 92d82fd2-cb48-4e03-d1e6-2a614651cb09
CSeq: 32010 REGISTER
Expires: 200
Content-Length: 0
Via: SIP/2.0/TCP
180.253.212.66:17589;rport;branch=z9hG4bK5M5v4xgLelotJ5NDt8hwPcuOWEFrz5Xu;ws-hacked=WSS
Server: Asterisk PBX 11.1.2
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Date: 14 Jan 2013 8:34:3 GMT;14

tsk_utils.js:110
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx
tsk_utils.js:110
==session event = connected call.htm:720
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=NOTIFY sip:10...@87.106.69.240:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.253.212.66;ws-src-port=17589;ws-src-proto=wss
SIP/2.0
Via: SIP/2.0/UDP 37.247.55.238:5060;branch=z9hG4bK2e1d552f
From: "asterisk"<sip:aste...@37.247.55.238>;tag=as17d9dded
To: <sip:10...@87.106.69.240:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.253.212.66;ws-src-port=17589;ws-src-proto=wss>
Contact: <sips:aste...@87.106.69.240:10060;transport=wss>
Call-ID: 4287547820718aed...@37.247.55.238:5060
CSeq: 102 NOTIFY
Content-Type: application/simple-message-summary
Content-Length: 93
Max-Forwards: 70
User-Agent: Asterisk PBX 11.1.2
Event: message-summary

Messages-Waiting: no
Message-Account: sip:aste...@37.247.55.238
Voice-Message: 0/0 (0/0)
tsk_utils.js:110
SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/UDP 37.247.55.238:5060;branch=z9hG4bK2e1d552f
From: "asterisk"<sip:aste...@37.247.55.238>;tag=as17d9dded
To: <sip:10...@87.106.69.240:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.253.212.66;ws-src-port=17589;ws-src-proto=wss>
Call-ID: 4287547820718aed...@37.247.55.238:5060
CSeq: 102 NOTIFY
Content-Length: 0

tsk_utils.js:110
State machine: c0000_Started_2_Outgoing_X_oINVITE tsk_utils.js:110
PeerConnectionClass = function RTCPeerConnection() { [native code] }
SessionDescriptionClass = function RTCSessionDescription() { [native
code] } IceCandidateClass = function RTCIceCandidate() { [native code]
} tsk_utils.js:110
==stack event = m_permission_requested call.htm:632
==session event = connecting call.htm:720
onGetUserMediaSuccess tsk_utils.js:110
createOffer tsk_utils.js:110
onCreateSdpSuccess tsk_utils.js:110
==stack event = m_permission_accepted call.htm:632
==session event = m_stream_audio_local_added call.htm:720
__on_state_change tsk_utils.js:110
onSetLocalDescriptionSuccess tsk_utils.js:110
onIceCandidate = closed tsk_utils.js:110
onIceCandidate = closed tsk_utils.js:110
onIceCandidate = closed tsk_utils.js:110
onIceCandidate = closed tsk_utils.js:110
onIceCandidate = closed tsk_utils.js:110
ICE GATHERING COMPLETED! tsk_utils.js:110
onIceGatheringCompleted tsk_utils.js:110
SEND: INVITE sip:fs_...@kuk.uk.to SIP/2.0
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bK7h3e4L3ewPlG2JcR2oLq4PCMZJTNV9Gs;rport
From: <sip:10...@kuk.uk.to>;tag=IzvZ6t59IW8feXDzoT6D
To: <sip:fs_...@kuk.uk.to>
Contact: "10...@kuk.uk.t"<sips:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;impi=1061;ha1=3a81dec123d272b95c91b890ae4488b2;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 6c7208f7-8d75-df47-d749-3d13ac2ec7b1
CSeq: 28060 INVITE
Content-Type: application/sdp
Content-Length: 1299
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.01.14
Organization: Doubango Telecom

v=0
o=- 4085513629 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
m=audio 17593 RTP/SAVPF 103 104 111 0 8 106 105 13 126
c=IN IP4 180.253.212.66
a=rtcp:17593 IN IP4 180.253.212.66
a=candidate:1788120350 1 udp 2113937151 10.0.11.151 63243 typ host generation 0
a=candidate:1788120350 1 udp 1677729535 180.253.212.66 17593 typ srflx
generation 0
a=candidate:1788120350 2 udp 2113937151 10.0.11.151 63243 typ host generation 0
a=candidate:1788120350 2 udp 1677729535 180.253.212.66 17593 typ srflx
generation 0
a=ice-ufrag:xzlBYAmO7p1LbbXy
a=ice-pwd:N/q4zWKtO9kDmrUJSalt2Z3q
a=ice-options:google-ice
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32
inline:Ng6Mi3zUUI9OJaLMpzdU8203BAUGQLrV0VUTwuxg
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:GLiqVMw60J6aLUdgj4n9monxqqKkwPweWjwZyLnl
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:111 opus/48000
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:126 telephone-event/8000
a=ssrc:2918753466 cname:8GunsWfUi5OvgFuW
a=ssrc:2918753466 msid:K0SzFBSastJUZIuGtM4ZIJY9BuA46agRrp8m a0
a=ssrc:2918753466 mslabel:K0SzFBSastJUZIuGtM4ZIJY9BuA46agRrp8m
a=ssrc:2918753466 label:K0SzFBSastJUZIuGtM4ZIJY9BuA46agRrp8ma0
tsk_utils.js:110
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;rport;branch=z9hG4bK7h3e4L3ewPlG2JcR2oLq4PCMZJTNV9Gs
From: <sip:10...@kuk.uk.to>;tag=IzvZ6t59IW8feXDzoT6D
To: <sip:fs_...@kuk.uk.to>
Call-ID: 6c7208f7-8d75-df47-d749-3d13ac2ec7b1
CSeq: 28060 INVITE
Content-Length: 0

tsk_utils.js:110
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:110
==session event = i_ao_request call.htm:720
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;rport;branch=z9hG4bK7h3e4L3ewPlG2JcR2oLq4PCMZJTNV9Gs
From: <sip:10...@kuk.uk.to>;tag=IzvZ6t59IW8feXDzoT6D
To: <sip:fs_...@kuk.uk.to>;tag=1358188197923
Contact: <sips:fs_...@87.106.69.240:10060;transport=wss;ws-src-ip=180.253.212.66;ws-src-port=17589;ws-src-proto=wss>
Call-ID: 6c7208f7-8d75-df47-d749-3d13ac2ec7b1
CSeq: 28060 INVITE
Content-Length: 0
Allow: ACK,BYE,CANCEL,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE

tsk_utils.js:110
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:110
==session event = i_ao_request call.htm:720
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 200 OK
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;rport;branch=z9hG4bK7h3e4L3ewPlG2JcR2oLq4PCMZJTNV9Gs
From: <sip:10...@kuk.uk.to>;tag=IzvZ6t59IW8feXDzoT6D
To: <sip:fs_...@kuk.uk.to>;tag=1358188197923
Contact: <sips:fs_...@87.106.69.240:10060;transport=wss;ws-src-ip=180.253.212.66;ws-src-port=17589;ws-src-proto=wss>
Call-ID: 6c7208f7-8d75-df47-d749-3d13ac2ec7b1
CSeq: 28060 INVITE
Content-Type: application/sdp
Content-Length: 707
Allow: ACK,BYE,CANCEL,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE

v=0
o=doubango 1983 678901 IN IP4 87.106.69.240
s=-
c=IN IP4 87.106.69.240
t=0 0
m=audio 43624 RTP/SAVPF 0 8
c=IN IP4 87.106.69.240
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=crypto:0 AES_CM_128_HMAC_SHA1_32
inline:OCBeTk5oz/qZE4BjEZB8a/kV5Pfti6Eh3GSvDb1O
a=sendrecv
a=rtcp-mux
a=ssrc:1209747078 cname:ldjWoB60jbyQlR6e
a=ssrc:1209747078 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:1209747078 label:Doubango
a=ice-ufrag:9bMizT6dbB64K9i
a=ice-pwd:CtRjCEnmOioVmzUYJf2gL
a=candidate:56mcuGazm 1 udp 2130706431 87.106.69.240 43624 typ host
a=candidate:srflx56mc 1 udp 1694498815 87.106.69.240 43624 typ srflx
raddr 87.106.69.240 rport 43624
tsk_utils.js:110
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE tsk_utils.js:110
setRemoteDescription(answer)
v=0
o=doubango 1983 678901 IN IP4 87.106.69.240
s=-
c=IN IP4 87.106.69.240
t=0 0
m=audio 43624 RTP/SAVPF 0 8
c=IN IP4 87.106.69.240
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=crypto:0 AES_CM_128_HMAC_SHA1_32
inline:OCBeTk5oz/qZE4BjEZB8a/kV5Pfti6Eh3GSvDb1O
a=sendrecv
a=rtcp-mux
a=ssrc:1209747078 cname:ldjWoB60jbyQlR6e
a=ssrc:1209747078 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:1209747078 label:Doubango
a=ice-ufrag:9bMizT6dbB64K9i
a=ice-pwd:CtRjCEnmOioVmzUYJf2gL
a=candidate:56mcuGazm 1 udp 2130706431 87.106.69.240 43624 typ host
a=candidate:srflx56mc 1 udp 1694498815 87.106.69.240 43624 typ srflx
raddr 87.106.69.240 rport 43624
tsk_utils.js:110
SEND: ACK sips:fs_...@87.106.69.240:10060;transport=wss;ws-src-ip=180.253.212.66;ws-src-port=17589;ws-src-proto=wss
SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKKzEHpLhLzUHQ3xeudXuP;rport
From: <sip:10...@kuk.uk.to>;tag=IzvZ6t59IW8feXDzoT6D
To: <sip:fs_...@kuk.uk.to>;tag=1358188197923
Contact: "10...@kuk.uk.t"<sips:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 6c7208f7-8d75-df47-d749-3d13ac2ec7b1
CSeq: 28060 ACK
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.01.14
Organization: Doubango Telecom

tsk_utils.js:110
__on_open tsk_utils.js:110
__on_state_change tsk_utils.js:110
__on_add_stream tsk_utils.js:110
onSetRemoteDescriptionSuccess tsk_utils.js:110
==session event = m_early_media call.htm:720
==session event = connected call.htm:720
==session event = m_stream_audio_remote_added call.htm:720
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 200 OK
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;rport;branch=z9hG4bK7h3e4L3ewPlG2JcR2oLq4PCMZJTNV9Gs
From: <sip:10...@kuk.uk.to>;tag=IzvZ6t59IW8feXDzoT6D
To: <sip:fs_...@kuk.uk.to>;tag=1358188197923
Contact: <sips:fs_...@87.106.69.240:10060;transport=wss;ws-src-ip=180.253.212.66;ws-src-port=17589;ws-src-proto=wss>
Call-ID: 6c7208f7-8d75-df47-d749-3d13ac2ec7b1
CSeq: 28060 INVITE
Content-Type: application/sdp
Content-Length: 707
Allow: ACK,BYE,CANCEL,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE

v=0
o=doubango 1983 678901 IN IP4 87.106.69.240
s=-
c=IN IP4 87.106.69.240
t=0 0
m=audio 43624 RTP/SAVPF 0 8
c=IN IP4 87.106.69.240
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=crypto:0 AES_CM_128_HMAC_SHA1_32
inline:OCBeTk5oz/qZE4BjEZB8a/kV5Pfti6Eh3GSvDb1O
a=sendrecv
a=rtcp-mux
a=ssrc:1209747078 cname:ldjWoB60jbyQlR6e
a=ssrc:1209747078 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:1209747078 label:Doubango
a=ice-ufrag:9bMizT6dbB64K9i
a=ice-pwd:CtRjCEnmOioVmzUYJf2gL
a=candidate:56mcuGazm 1 udp 2130706431 87.106.69.240 43624 typ host
a=candidate:srflx56mc 1 udp 1694498815 87.106.69.240 43624 typ srflx
raddr 87.106.69.240 rport 43624
tsk_utils.js:110
State machine: x0000_Any_2_Any_X_i2xxINVITE tsk_utils.js:110
Remote offer has not changed tsk_utils.js:116
SEND: ACK sips:fs_...@87.106.69.240:10060;transport=wss;ws-src-ip=180.253.212.66;ws-src-port=17589;ws-src-proto=wss
SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKM2qI77g1TUVG1TU39Dbo;rport
From: <sip:10...@kuk.uk.to>;tag=IzvZ6t59IW8feXDzoT6D
To: <sip:fs_...@kuk.uk.to>;tag=1358188197923
Contact: "10...@kuk.uk.t"<sips:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 6c7208f7-8d75-df47-d749-3d13ac2ec7b1
CSeq: 28060 ACK
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.01.14
Organization: Doubango Telecom

tsk_utils.js:110
onIceCandidate = closed tsk_utils.js:110
onIceCandidate = closed tsk_utils.js:110
ICE GATHERING COMPLETED! tsk_utils.js:110
onIceGatheringCompleted tsk_utils.js:110
onIceGatheringCompleted but no local sdp request is pending tsk_utils.js:116
onIceCandidate = closed tsk_utils.js:110
ICE GATHERING COMPLETED! tsk_utils.js:110
onIceGatheringCompleted tsk_utils.js:110
onIceGatheringCompleted but no local sdp request is pending tsk_utils.js:116
State machine: x0000_Any_2_Trying_X_oBYE tsk_utils.js:110
SEND: BYE sips:fs_...@87.106.69.240:10060;transport=wss;ws-src-ip=180.253.212.66;ws-src-port=17589;ws-src-proto=wss
SIP/2.0
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bKATIVfzMJyOLUSePWbHAm2NGzkqk5hO4c;rport
From: <sip:10...@kuk.uk.to>;tag=IzvZ6t59IW8feXDzoT6D
To: <sip:fs_...@kuk.uk.to>;tag=1358188197923
Call-ID: 6c7208f7-8d75-df47-d749-3d13ac2ec7b1
CSeq: 28061 BYE
Content-Length: 0
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/2.0 200 OK
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;rport;branch=z9hG4bKATIVfzMJyOLUSePWbHAm2NGzkqk5hO4c
From: <sip:10...@kuk.uk.to>;tag=IzvZ6t59IW8feXDzoT6D
To: <sip:fs_...@kuk.uk.to>;tag=1358188197923
Contact: <sips:fs_...@87.106.69.240:10060;transport=wss;ws-src-ip=180.253.212.66;ws-src-port=17589;ws-src-proto=wss>
Call-ID: 6c7208f7-8d75-df47-d749-3d13ac2ec7b1
CSeq: 28061 BYE
Content-Length: 0

tsk_utils.js:110
State machine: x0000_Any_2_Terminated_X_i2xxBYE tsk_utils.js:110
=== INVITE Dialog terminated === tsk_utils.js:110
PeerConnection::stop() tsk_utils.js:110
==session event = terminated call.htm:720
The FSM is in the final state

--
Iwan Budi Kusnanto

Mamadou

unread,
Jan 14, 2013, 6:50:26 AM1/14/13
to discuss-webrtc
Please provide feedbacks on our mailing list to avoid more spamming :)

On Jan 14, 9:37 am, Iwan Budi Kusnanto <i...@labhijau.net> wrote:
> impi='1061', impu='<sip:1...@kuk.uk.to>' tsk_utils.js:110
> Connecting to 'wss://sipml5.org:10062' tsk_utils.js:110
> ==stack event = starting call.htm:632
> __tsip_transport_ws_onopen tsk_utils.js:110
> ==stack event = started call.htm:632
> State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister
> tsk_utils.js:110
> SEND: REGISTER sip:kuk.uk.to SIP/2.0
> Via: SIP/2.0/WSS
> df7jal23ls0d.invalid;branch=z9hG4bK6z6ZCcIsqg7rQXL7IcCqlRtge1eGw1E8;rport
> From: <sip:1...@kuk.uk.to>;tag=UyU5RaiAzbxnnRBf4zTP
> To: <sip:1...@kuk.uk.to>
> Contact: "1...@kuk.uk.t"<sips:1...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;expires=200;+g.oma. sip-im;+audio;language="en,fr"
> Call-ID: 92d82fd2-cb48-4e03-d1e6-2a614651cb09
> CSeq: 32009 REGISTER
> Content-Length: 0
> Max-Forwards: 70
> User-Agent: IM-client/OMA1.0 sipML5-v1.2013.01.14
> Organization: Doubango Telecom
> Supported: path
>
>  tsk_utils.js:110
> ==session event = connecting call.htm:720
> ==session event = sent_request call.htm:720
> __tsip_transport_ws_onmessage tsk_utils.js:110
> recv=SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 87.106.69.240:10060;rport=10060;received=87.106.69.240;branch=z9hG4bK6z6ZCc Isqg7rQXL7IcCqlRtge1eGw1E8
> From: <sip:1...@kuk.uk.to>;tag=UyU5RaiAzbxnnRBf4zTP
> To: <sip:1...@kuk.uk.to>;tag=as008c7451
> Call-ID: 92d82fd2-cb48-4e03-d1e6-2a614651cb09
> CSeq: 32009 REGISTER
> Content-Length: 0
> Via: SIP/2.0/TCP
> 180.253.212.66:17589;rport;branch=z9hG4bK6z6ZCcIsqg7rQXL7IcCqlRtge1eGw1E8;w s-hacked=WSS
> Server: Asterisk PBX 11.1.2
> Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
> Supported: replaces,timer
> WWW-Authenticate: Digest
> realm="kuk.uk.to",nonce="62c6a6b4",stale=FALSE,algorithm=MD5
>
>  tsk_utils.js:110
> State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
> tsk_utils.js:110
> SEND: REGISTER sip:kuk.uk.to SIP/2.0
> Via: SIP/2.0/WSS
> df7jal23ls0d.invalid;branch=z9hG4bK5M5v4xgLelotJ5NDt8hwPcuOWEFrz5Xu;rport
> From: <sip:1...@kuk.uk.to>;tag=UyU5RaiAzbxnnRBf4zTP
> To: <sip:1...@kuk.uk.to>
> Contact: "1...@kuk.uk.t"<sips:1...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;expires=200;+g.oma. sip-im;+audio;language="en,fr"
> Call-ID: 92d82fd2-cb48-4e03-d1e6-2a614651cb09
> CSeq: 32010 REGISTER
> Content-Length: 0
> Max-Forwards: 70
> Authorization: Digest
> username="1061",realm="kuk.uk.to",nonce="62c6a6b4",uri="sip:kuk.uk.to",resp onse="2a8ff6db3255129083aa33df0cf039e9",algorithm=MD5
> User-Agent: IM-client/OMA1.0 sipML5-v1.2013.01.14
> Organization: Doubango Telecom
> Supported: path
>
>  tsk_utils.js:110
> ==session event = sent_request call.htm:720
> __tsip_transport_ws_onmessage tsk_utils.js:110
> recv=SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 87.106.69.240:10060;rport=10060;received=87.106.69.240;branch=z9hG4bK5M5v4x gLelotJ5NDt8hwPcuOWEFrz5Xu
> From: <sip:1...@kuk.uk.to>;tag=UyU5RaiAzbxnnRBf4zTP
> To: <sip:1...@kuk.uk.to>;tag=as008c7451
> Contact: <sip:1...@87.106.69.240:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.253.212 .66;ws-src-port=17589;ws-src-proto=wss>;expires=200
> Call-ID: 92d82fd2-cb48-4e03-d1e6-2a614651cb09
> CSeq: 32010 REGISTER
> Expires: 200
> Content-Length: 0
> Via: SIP/2.0/TCP
> 180.253.212.66:17589;rport;branch=z9hG4bK5M5v4xgLelotJ5NDt8hwPcuOWEFrz5Xu;w s-hacked=WSS
> Server: Asterisk PBX 11.1.2
> Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
> Supported: replaces,timer
> Date: 14 Jan 2013 8:34:3 GMT;14
>
>  tsk_utils.js:110
> State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx
> tsk_utils.js:110
> ==session event = connected call.htm:720
> __tsip_transport_ws_onmessage tsk_utils.js:110
> recv=NOTIFY sip:1...@87.106.69.240:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.253.212 .66;ws-src-port=17589;ws-src-proto=wss
> SIP/2.0
> Via: SIP/2.0/UDP 37.247.55.238:5060;branch=z9hG4bK2e1d552f
> From: "asterisk"<sip:aster...@37.247.55.238>;tag=as17d9dded
> To: <sip:1...@87.106.69.240:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.253.212 .66;ws-src-port=17589;ws-src-proto=wss>
> Contact: <sips:aster...@87.106.69.240:10060;transport=wss>
> Call-ID: 4287547820718aed2ebb73990c3b3...@37.247.55.238:5060
> CSeq: 102 NOTIFY
> Content-Type: application/simple-message-summary
> Content-Length: 93
> Max-Forwards: 70
> User-Agent: Asterisk PBX 11.1.2
> Event: message-summary
>
> Messages-Waiting: no
> Message-Account: sip:aster...@37.247.55.238
> Voice-Message: 0/0 (0/0)
>  tsk_utils.js:110
> SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist
> Via: SIP/2.0/UDP 37.247.55.238:5060;branch=z9hG4bK2e1d552f
> From: "asterisk"<sip:aster...@37.247.55.238>;tag=as17d9dded
> To: <sip:1...@87.106.69.240:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=180.253.212 .66;ws-src-port=17589;ws-src-proto=wss>
> Call-ID: 4287547820718aed2ebb73990c3b3...@37.247.55.238:5060
> From: <sip:1...@kuk.uk.to>;tag=IzvZ6t59IW8feXDzoT6D
> To: <sip:fs_...@kuk.uk.to>
> Contact: "1...@kuk.uk.t"<sips:1...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;impi=1061;ha1=3a81d ec123d272b95c91b890ae4488b2;+g.oma.sip-im;+sip.ice;language="en,fr"
> From: <sip:1...@kuk.uk.to>;tag=IzvZ6t59IW8feXDzoT6D
> To: <sip:fs_...@kuk.uk.to>
> Call-ID: 6c7208f7-8d75-df47-d749-3d13ac2ec7b1
> CSeq: 28060 INVITE
> Content-Length: 0
>
>  tsk_utils.js:110
> State machine: ...
>
> read more »

Jesús Leganés Combarro

unread,
Jan 16, 2013, 3:01:34 AM1/16/13
to discuss...@googlegroups.com

Have you tested it with data channels?

+1, if the RTCWeb breaker works with DataChannels I'm interested to be a betatester :-) 
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