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Failed to set remote offer sdp: Called with SDP without DTLS
Hello,
When I answer to a call I got this error
I'm using asterisk server side
Any ideas ?
Thanks for your help !
SIPjs can't get audio on chrome browser
Hi all,
I have the problem with video/audio call with SIPjs client.
I use Kazoo system and create SIP account to make call.
First, I make call with audio (PCMU), it work ok.
After that, I enable video codec (VP8) for SIP user. Then make audio call (no video). I face the problem:
- I can not listen the audio from callee (receiver)
- After that, I debug on Freeswitch and see:
- Caller make audio call with audio codec (PCMU) on SDP
- Freeswitch send Invite to callee with both audio codec (PCMU) and video codec (VP8). Because Kazoo see the callee support both video and audio codec.
- So that, callee make video + audio stream with Freeswitch.
Some cases, I have tested:
- Chrome call Chrome, Firefox call Chrome: Callee can't listen
- Chrome call Firefox, Firefox call Firefox: Callee can listen ok
=> When call to Chrome, I see chrome://webrtc-internals/ , the client can send/receive all packets but the audioOutputLevel of recevicer = 0 .
rfc2833 support for dtmf in sip.js 0.17.x
I didn't find any option to enable support of rfc2833 in-band dtmf support in latest sip.js version 0.17.x
While it was available in earlier version like it was till 0.15.x but I didn't find option to configure same. Does anyone have idea about how to or is there any reason to remove it?
Session's dtmf event is incorrectly documented
http://sipjs.com/api/0.6.0/session/#dtmf
.on('dtmf', function(dtmf, request) {})
Source code:
Session/DTMF.js: this.owner.emit('dtmf', this.request, this);
Session/DTMF.js: this.owner.emit('dtmf', request, this);
I think that the code is correct and the documentation is wrong, because putting the request/response first seems to be the convention in your library.
Update guides to use v0.7.0
The guides should be updated to the latest version. There are some breaking changes between 0.6.x and 0.7.0 and the guides could mislead people.
For example, the "Receive a Call" guide. We must change
userAgent.on('invite', function (session) {
session.accept({
media: {
render: {
remote: {
video: document.getElementById('remoteVideo')
},
local: {
video: document.getElementById('localVideo')
}
}
}
});
});to
userAgent.on('invite', function (session) {
session.accept({
media: {
render: {
remote: document.getElementById('remoteVideo'),
local: document.getElementById('localVideo')
}
}
});
});There are probably other guides that need to be updated as well.
Refer guide is out of date
It uses the referred listener instead of refer, and does not have the new followRefer function mentioned. Could probably use some TLC.
via rport parse failed.
I tried to get rport param from Via, but I found that this param could not be parsed in SIP.Grammar.parse.
my sip.js version is 0.12.0. latest version chrome browser.
parse rport code in sip.js like this:
11219 line：
"rport",
peg$literalExpectation("rport", true),
function () {
if (typeof response_port !== 'undefined')
options.data.rport = response_port.join('');
},
I can't find response_port param in sip.js. so rport is always undefined. And I try to get the split rport, but i get this params is empty.
I want to know if rport has other ways to get it in sip.js? Or there is a bug with this code.
this my sip response.
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 114.116.146.102;branch=z9hG4bK5740053;rport=51734;received=101.71.41.233
To: "nhua_test1" sip:[email protected]:51966;tag=b1a36eeb7e08ea8ab155ea66a70e7fe0.436f
From: "nhua_test1" sip:[email protected]:51966;tag=ae8hpprt2o
Call-ID: mgrbq2giipp4etkms1uvbl
CSeq: 6235 REGISTER
WWW-Authenticate: Digest realm="114.116.146.102", nonce="XGukIlxrovbP8Ei8CK7hmpz19JctW/y+"
Server: Vanish 1.0.1
Content-Length: 0
Cordova Guide does not mention phonertcmediahandler.js
Even though it is not ready for primetime yet, the guide is missing an explanation of phonertcmediahandler.js which is a necessary piece to get sip.js working with cordova and phonertc.
UA config `reliable` option is undocumented
rel100 was deprecated and replaced by reliable, but the docs don't say anything about reliable
Unable to Render v0.7.0 with Asterisk
I seem to only be able to hear audio only by passing an audio element not an options object on my invite. Am I missing something or is there an issue with the options object?
So this doesn't work:
var media_options = {
media: {
constraints: {
audio: true,
video: false
},
render: {
remote: document.getElementById('remoteAudio'),
local: document.getElementById('localAudio')
}
}
};
var session = UA.invite(document.getElementById('destination').value,media_options);
But this does:
var session = UA.invite(document.getElementById('destination').value,document.getElementById('remoteAudio'));
Can't receive DTMF tones from SIP trunk
Hi!
I have a problem receiving DTMF tones from a SIP trunk (PSTN). I can hear the tones, but .on('dtmf') is not trigged. Sending DTMF tones from sipjs workes great. My test code is:
session.on('dtmf', function(response, dtmf) {
console.log('Tone:', dtmf);
});
Can someone help me with this issue?
(I'm running latest version downloaded with bower)
Allow using a custom console
Some JS frameworks offer their own wrapper around window.console (i.e. Angular's $log).
It would be very convenient to be able to tell SIP.js which console object to use when trying to print messages.
clarify documentation: new SIP.UA calls start()
Instantiating a SIP.UA seems to implicitly call .start(), but there is currently no mention of this in the documentation.
Simple Guide has a broken link to Simple User
Describe the bug
https://sipjs.com/guides/simple/ has a link in "You should consider upgrading to the Simple User on the latest SIP.js Version." but the link 404's. It is currently https://sipjs.com/simple-user
Logs
Please include a gist or attached file with full logs. If this is not included, the first response will likely be requesting them.
To Reproduce (if possible)
Steps to reproduce the behavior:
Go to 'https://sipjs.com/guides/simple/'
Click on 'Simple User' within the Disclaimer section
See 404 error
Expected behavior
the correct (unknown so far) Simple User guide
Observed behavior
404 error.
Environment Information
Additional context
I expect the correct link is: https://sipjs.com/guides/simple-user/
How do I get the same call id from freeswitch
I am trying to identify calls from Freeswitch using a socket event connection.
the idea is I want to log a call info into a DB that is independent of Freeswitch using an event socket connection, then associate it on the client side using some unique call id. The reason is I want to be able to use data in my application to log or manipulate it in many ways when a user misses, answers etc, a particular call since that call may never be intercepted by SIPjs (user might be offline).
When a call comes in, the following Nodejs script intercepts an event and is given the call information:
const esl = require('modesl');
const conn = new esl.Connection('192.168.43.5', 8021, 'ClueCon', function (ev) {
conn.on('esl::**', function (evt) {
console.debug(`index.js-50: () - evt: %o`, evt);
});
});However, the (_.find(evt.headers, {name: 'variable_sip_call_id'})).value in the headers of that event is different from the call id in sipjs InviteServerContext.request.callId (I just can't find the same value anywhere, I have checked other variables in the header). Here is an example:
Freeswitch
{
name: 'variable_sip_call_id',
value: '101921M2JjZTgyYzQ5YzE5ZDI0MWI3MmJlNTRkNWYzYzhjM2I'
}SIPjs
{ callId: "a9590375-7fd2-1238-5bbd-f04da2cc571c" }How do I associate this call?
Early Media and Hangup
I am using SIPJS for my webrtc client ,When i try to make call pstn, i cant able to hear any early media and if b leg (pstn) disconnect call ,A-leg(sipjs) did't receive bye message.
if i use this same scenario in my voipfone(xlite,zoiper) works fine.
SIPJS Demo Call hangup automatically
SIPJS(0.7.1) demo phone call(audio/video) hangup automatically after 60 seconds,using with freeswitch
google stun server hardcoded
when the nachines are not connected to the internet it takes a lot of time (30 second timeut) to make the call. i have checked several times but in configuration the stun servers value was an empty array.
so i checked your source code and found you have hardcoded the google stun server as default.
the only way we had to run the application was to remove it (editing sip.js file) which is vey bad practice. so i think it's a bug and you may leave it to user to insert google stun server(a matter of documentation and informing user).
note i have tried to set the default value to empty string but no success. the only way it worked ws to set iceservers parameter to [] when calling RTCpeerConnection.
Overriding Media Handler
We should have a guide on how to override the default WebRTC media handler and create your own.
Option to stop sending PRACKs
In some systems PRACKs are not supported and lead to 4xx errors.
PRACKs are covered by a separate RFC https://www.ietf.org/rfc/rfc3262.txt and are not a part of SIP RFC 3261. This RFC is focused on guaranteed establishing of a media session before 200OK. In RingCentral system there are no scenarios, which require a media session before the call is established. Moreover, most of SIP flows are built in an way where a leg is Answered before it any media is played.
With PRACKs calls are successfully established, but users get 4xx errors so the logs are not clean.
SIP.Parser is orphan
http://sipjs.com/api/0.6.0/parser/
This page does not appear to link from anywhere else.
incomingMessage -> sipMessage
There are a bunch of broken links to incomingMessage docs, but now the file has been combined with many other objects in a file called sipMessage. Let's fix those links.
RTCIceChecking Timeout Triggered after 5000 milliseconds
I am having a strange problem and that only occurs when I make an outbound call. I tried using the SIP.js demo phone. When I dial an external number and the called party answers the call within 5 seconds then the media is established end to end but if the called party answers the call after 5 seconds then media from browser to asterisk is not established and its a one way audio call. Asterisk however plays MOH and DTMF.
Since the browser is running on https (self signed cert) the tcpdump on the server only traces the outside leg. Upon comparison I found no difference in signalling / SDP packets. Same was done at the client end browser. Signalling / SDP etc remains the same (or I wasn't able to find out) but SIP.js prints a log the says the following
------- in case of 1 way audio -------------
sip.js:36 Thu Aug 31 2017 19:02:11 GMT+0500 (Pakistan Standard Time) | sip.invitecontext.mediahandler | 5ht89pfuco7uopuuvnfvr6vgdksft0 | stream added: default
sip.js:36 Thu Aug 31 2017 19:02:16 GMT+0500 (Pakistan Standard Time) | sip.invitecontext.mediahandler | 5ht89pfuco7uopuuvnfvr6vgdksft0 | RTCIceChecking Timeout Triggered after 5000 milliseconds
sip.js:36 Thu Aug 31 2017 19:02:20 GMT+0500 (Pakistan Standard Time) | sip.transaction.ict | z9hG4bK3649713 | Timer B expired for INVITE client transaction z9hG4bK3649713
sip.js:36 Thu Aug 31 2017 19:02:28 GMT+0500 (Pakistan Standard Time) | sip.transport | received WebSocket text message:
------- in case of successful call -------------
sip.js:36 Thu Aug 31 2017 18:43:42 GMT+0500 (Pakistan Standard Time) | sip.invitecontext.mediahandler | qotki1742sonfr93nuem74m4k726sf | stream added: default
sip.js:36 Thu Aug 31 2017 18:43:45 GMT+0500 (Pakistan Standard Time) | sip.transport | received WebSocket text message:
I am attaching the browser SIP Logs from the console if anyone can help me out.
no-voice10.11.11.75-1504188160056.txt
10.11.11.75-1504187045459.txt
I tried playing with iceCheckingTimeout but nothing helped. My UA looks like
config : {
password : user.Pass,
displayName : user.Display,
uri : 'sip:'+user.User+'@'+user.Realm,
wsServers : user.WSServer,
// stunServers : [],
// usePreloadedRoute : true,
registerExpires : 600,
traceSip : true,
hackWssInTransport: true,
hackIpInContact: true,
log : {
level : 3,
},
rtcpMuxPolicy: "negotiate",
iceCheckingTimeout: 500 // iceCheckingTimeout: 60000
},
Any other logs required please let me know.
Thank you
Habib
Chrome error when answering call with two audio streams
The following SDP is on an inbound INVITE to sip.js 0.8.3. When a session.accept() is done in Chrome, it throws an error. This works fine in Firefox.
It looks like both audio streams are being added. When it comes to adding the second one, Chrome rejects it.
If Chrome can't handle multiple streams in a single PeerConnection, perhaps one should be rejected or multiple PeerConnections should be used?
SDP
o=SBC 1511241829 1511241830 IN IP4 34.240.105.141
s=SBC
c=IN IP4 34.240.105.141
t=0 0
m=audio 20270 RTP/SAVPF 102 8 0 18 9 103 101
a=rtpmap:102 opus/48000/2
a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; ptime=20; minptime=10; maxptime=40
a=fmtp:18 annexb=no
a=rtpmap:103 telephone-event/48000
a=fmtp:103 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=rtcp:20271
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:SGzE63XDPZrbV4MpX8ryA+idPjVFWGOFaWsDNvE9
a=setup:actpass
a=fingerprint:sha-1 67:70:92:C1:38:97:BA:0B:9A:F7:41:28:28:BE:A1:83:FD:D8:01:BB
a=ice-ufrag:5Sdhvqw3
a=ice-pwd:ov7pyAg9oA2KK3bc3y8GtuaSSO
a=candidate:QUveaoD2tueydfZ1 1 UDP 2130706431 34.240.105.141 20270 typ host
a=candidate:QUveaoD2tueydfZ1 2 UDP 2130706430 34.240.105.141 20271 typ host
m=audio 20306 RTP/SAVPF 102 8 0 18 9 103 101
a=rtpmap:102 opus/48000/2
a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; ptime=20; minptime=10; maxptime=40
a=fmtp:18 annexb=no
a=rtpmap:103 telephone-event/48000
a=fmtp:103 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=rtcp:20307
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:tvwADoGGWZCgybFlyiU5pSfKayBLsSx/yZNvw1Rb
a=setup:actpass
a=fingerprint:sha-1 67:70:92:C1:38:97:BA:0B:9A:F7:41:28:28:BE:A1:83:FD:D8:01:BB
a=ice-ufrag:i7cO4OrT
a=ice-pwd:dJaz4LuZQziYzeyifzztyMa5BE
a=candidate:QUveaoD2tueydfZ1 1 UDP 2130706431 34.240.105.141 20306 typ host
a=candidate:QUveaoD2tueydfZ1 2 UDP 2130706430 34.240.105.141 20307 typ host
Error inc stack trace
LoggerFactory.print @ sip-0.8.3.js:783
LoggerFactory.(anonymous function) @ sip-0.8.3.js:800
Logger.(anonymous function) @ sip-0.8.3.js:794
setRemoteDescriptionError @ sip-0.8.3.js:8735
Promise rejected (async)
setDescription @ sip-0.8.3.js:8733
accept @ sip-0.8.3.js:5709
(anonymous) @ ua.js:66
dispatch @ jquery-3.2.1.min.js:3
q.handle @ jquery-3.2.1.min.js:3
Session's "dialog" event is undocumented
This was introduced in onsip/SIP.js@a65fde7, but is not mentioned at http://sipjs.com/api/0.7.0/session/#events
See onsip/SIP.js#275 for an example of a problem that can be mitigated by using the "dialog" event.
Cordova with SIPJS
Hello there !
I was using sipjs with cordova on android 6 and everything was working well !
I'm trying now sipjs with android 8.1. The app was correctyl ringing but I've no sound and micro was not working.
This log are taked directly from Android studio:
2021-11-12 18:25:51.290 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:51 GMT+0100 (heure normale d’Europe centrale) | sip.Transport | Received WebSocket text message:
INVITE sip:[email protected]:44514;transport=ws SIP/2.0
Via: SIP/2.0/WS 127.0.0.1:8088;rport;branch=z9hG4bKPja7de5ada-75f8-480e-9bfe-6d5d4bb5db75;alias
From: <sip:1001@XXXXXXXX>;tag=ba5942c8-e85c-4993-a708-5a16c8442d9a
To: <sip:[email protected]>
Contact: <sip:asterisk@XXXXXXXX:5060;transport=ws>
Call-ID: 480a0aac-9522-448a-9b3a-0aa63a83df32
CSeq: 16759 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-16-89cf7899be
Content-Type: application/sdp
Content-Length: 1093
v=0
o=- 2067970350 2067970350 IN IP4 10.11.72.99
s=Asterisk
c=IN IP4 10.11.72.99
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 15080 UDP/TLS/RTP/SAVPF 0 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 6D:44:00:93:35:F3:67:32:82:A2:BD:0A:0A:3E:35:8A:05:8A:B6:7A:33:BD:24:55:3C:8F:23:2D:71:DF:6A:1B
a=ice-ufrag:45363e373092d3f04854ff9418d59f01
a=ice-pwd:571182fd53355b19719950663f9785c4
a=candidate:Ha0b4863 1 UDP 2130706431 10.11.72.99 15080 typ host
a=candidate:Hc0a80263 1 UDP 2130706431 192.168.2.99 15080 typ host
a=candidate:H1a000e1e 1 UDP 2130706431 26.0.14.30 15080 typ host
a=candidate:H43c8d275 1 UDP 2130706431 fe80::20d:b9ff:fe58:c0fa 15080 typ host
a=candidate:H5f685279 1 UDP 2130706431 fe80::3535:5177:4b67:fb5c 15080 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp-mux
a=ssrc:719363457 cname:6cdeebe4-a265-4636-9fb0-17762c6b96b4
a=msid:2e1bd094-1391-43b2-afc9-17d6857b6e99 74804d64-22aa-4c41-9db2-8a6a663d6ccc
a=rtcp-fb:* transport-cc
a=mid:audio-0
2021-11-12 18:25:51.291 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:51 GMT+0100 (heure normale d’Europe centrale) | sip.Transport | Received WebSocket text message:
INVITE sip:[email protected]:44514;transport=ws SIP/2.0
Via: SIP/2.0/WS 127.0.0.1:8088;rport;branch=z9hG4bKPja7de5ada-75f8-480e-9bfe-6d5d4bb5db75;alias
From: <sip:1001@XXXXXXXX>;tag=ba5942c8-e85c-4993-a708-5a16c8442d9a
To: <sip:[email protected]>
Contact: <sip:asterisk@XXXXXXXX:5060;transport=ws>
Call-ID: 480a0aac-9522-448a-9b3a-0aa63a83df32
CSeq: 16759 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-16-89cf7899be
Content-Type: application/sdp
Content-Length: 1093
v=0
o=- 2067970350 2067970350 IN IP4 10.11.72.99
s=Asterisk
c=IN IP4 10.11.72.99
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 15080 UDP/TLS/RTP/SAVPF 0 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 6D:44:00:93:35:F3:67:32:82:A2:BD:0A:0A:3E:35:8A:05:8A:B6:7A:33:BD:24:55:3C:8F:23:2D:71:DF:6A:1B
a=ice-ufrag:45363e373092d3f04854ff9418d59f01
a=ice-pwd:571182fd53355b19719950663f9785c4
a=candidate:Ha0b4863 1 UDP 2130706431 10.11.72.99 15080 typ host
a=candidate:Hc0a80263 1 UDP 2130706431 192.168.2.99 15080 typ host
a=candidate:H1a000e1e 1 UDP 2130706431 26.0.14.30 15080 typ host
a=candidate:H43c8d275 1 UDP 2130706431 fe80::20d:b9ff:fe58:c0fa 15080 typ host
a=candidate:H5f685279 1 UDP 2130706431 fe80::3535:5177:4b67:fb5c 15080 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp-mux
a=ssrc:719363457 cname:6cdeebe4-a265-4636-9fb0-17762c6b96b4
a=msid:2e1bd094-1391-43b2-afc9-17d6857b6e99 74804d64-22aa-4c41-9db2-8a6a663d6ccc
a=rtcp-fb:* transport-cc
a=mid:audio-0
", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:51.359 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:51 GMT+0100 (heure normale d’Europe centrale) | sip.Transport | Sending WebSocket message:
SIP/2.0 100 Trying
Via: SIP/2.0/WS 127.0.0.1:8088;rport;branch=z9hG4bKPja7de5ada-75f8-480e-9bfe-6d5d4bb5db75;alias
From: <sip:1001@XXXXXXXX>;tag=ba5942c8-e85c-4993-a708-5a16c8442d9a
To: <sip:[email protected]>
CSeq: 16759 INVITE
Call-ID: 480a0aac-9522-448a-9b3a-0aa63a83df32
Supported: outbound
User-Agent: SIP.js/0.16.0
Content-Length: 0
2021-11-12 18:25:51.359 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:51 GMT+0100 (heure normale d’Europe centrale) | sip.Transport | Sending WebSocket message:
SIP/2.0 100 Trying
Via: SIP/2.0/WS 127.0.0.1:8088;rport;branch=z9hG4bKPja7de5ada-75f8-480e-9bfe-6d5d4bb5db75;alias
From: <sip:1001@XXXXXXXX>;tag=ba5942c8-e85c-4993-a708-5a16c8442d9a
To: <sip:[email protected]>
CSeq: 16759 INVITE
Call-ID: 480a0aac-9522-448a-9b3a-0aa63a83df32
Supported: outbound
User-Agent: SIP.js/0.16.0
Content-Length: 0
", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:51.360 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:51 GMT+0100 (heure normale d’Europe centrale) | sip.Invitation | Invitation.progress
2021-11-12 18:25:51.360 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:51 GMT+0100 (heure normale d’Europe centrale) | sip.Invitation | Invitation.progress", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:51.360 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:51 GMT+0100 (heure normale d’Europe centrale) | sip.invite-dialog | INVITE dialog 480a0aac-9522-448a-9b3a-0aa63a83df3202r8ufl7kiba5942c8-e85c-4993-a708-5a16c8442d9a constructed
2021-11-12 18:25:51.360 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:51 GMT+0100 (heure normale d’Europe centrale) | sip.invite-dialog | INVITE dialog 480a0aac-9522-448a-9b3a-0aa63a83df3202r8ufl7kiba5942c8-e85c-4993-a708-5a16c8442d9a constructed", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:51.361 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:51 GMT+0100 (heure normale d’Europe centrale) | sip.Transport | Sending WebSocket message:
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 127.0.0.1:8088;rport;branch=z9hG4bKPja7de5ada-75f8-480e-9bfe-6d5d4bb5db75;alias
From: <sip:1001@XXXXXXXX>;tag=ba5942c8-e85c-4993-a708-5a16c8442d9a
To: <sip:[email protected]>;tag=02r8ufl7ki
CSeq: 16759 INVITE
Call-ID: 480a0aac-9522-448a-9b3a-0aa63a83df32
Supported: outbound
User-Agent: SIP.js/0.16.0
Contact: <sip:[email protected];transport=ws>
Content-Length: 0
2021-11-12 18:25:51.361 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:51 GMT+0100 (heure normale d’Europe centrale) | sip.Transport | Sending WebSocket message:
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 127.0.0.1:8088;rport;branch=z9hG4bKPja7de5ada-75f8-480e-9bfe-6d5d4bb5db75;alias
From: <sip:1001@XXXXXXXX>;tag=ba5942c8-e85c-4993-a708-5a16c8442d9a
To: <sip:[email protected]>;tag=02r8ufl7ki
CSeq: 16759 INVITE
Call-ID: 480a0aac-9522-448a-9b3a-0aa63a83df32
Supported: outbound
User-Agent: SIP.js/0.16.0
Contact: <sip:[email protected];transport=ws>
Content-Length: 0
", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:51.362 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:51 GMT+0100 (heure normale d’Europe centrale) | sip.SimpleUser | [Anonymous] Received INVITE
2021-11-12 18:25:51.362 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:51 GMT+0100 (heure normale d’Europe centrale) | sip.SimpleUser | [Anonymous] Received INVITE", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:51.362 W/AudioManager: Use of stream types is deprecated for operations other than volume control
2021-11-12 18:25:51.363 W/AudioManager: See the documentation of requestAudioFocus() for what to use instead with android.media.AudioAttributes to qualify your playback use case
2021-11-12 18:25:51.363 13713-14416/com.weble.shs I/AAudio: AAudioStreamBuilder_openStream() called ----------------------------------------
2021-11-12 18:25:51.363 13713-14416/com.weble.shs I/AudioStreamBuilder: rate = 44100, channels = 2, format = 5, sharing = SH, dir = OUTPUT
2021-11-12 18:25:51.363 13713-14416/com.weble.shs I/AudioStreamBuilder: device = 0, sessionId = -1, perfMode = 11, callback: ON with frames = 4104
2021-11-12 18:25:51.363 13713-14416/com.weble.shs I/AudioStreamBuilder: usage = 1, contentType = 0, inputPreset = 0, allowedCapturePolicy = 0
2021-11-12 18:25:51.363 13713-14416/com.weble.shs D/AudioStreamBuilder: build() MMAP not available because AAUDIO_PERFORMANCE_MODE_LOW_LATENCY not used.
2021-11-12 18:25:51.364 13713-14416/com.weble.shs D/AudioStreamTrack: open(), request notificationFrames = 4104, frameCount = 0
2021-11-12 18:25:51.368 D/SystemWebChromeClient: cdvfile://localhost/persistent/min.js: Line 1 : GOTO 1
2021-11-12 18:25:51.368 I/chromium: [INFO:CONSOLE(1)] "GOTO 1", source: cdvfile://localhost/persistent/min.js (1)
2021-11-12 18:25:51.368 D/SystemWebChromeClient: cdvfile://localhost/persistent/min.js: Line 1 : /app11/LayoutFlexTitle/Content/TabGrid/Content/grid translateTo Translate on same index !
2021-11-12 18:25:51.368 I/chromium: [INFO:CONSOLE(1)] "/app11/LayoutFlexTitle/Content/TabGrid/Content/grid translateTo Translate on same index !", source: cdvfile://localhost/persistent/min.js (1)
2021-11-12 18:25:51.369 13713-14416/com.weble.shs D/AudioTrack: setVolume(1.000000, 1.000000) pid : 13713
2021-11-12 18:25:51.371 13713-14416/com.weble.shs I/AAudio: AAudioStreamBuilder_openStream() returns 0 = AAUDIO_OK for s#7 ----------------
2021-11-12 18:25:51.371 13713-14416/com.weble.shs D/AAudio: AAudioStream_requestStart(s#7) called --------------
2021-11-12 18:25:51.378 13713-14365/com.weble.shs D/AudioStreamLegacy: onAudioDeviceUpdate() devId 3 => 3
2021-11-12 18:25:51.378 13713-14416/com.weble.shs D/AAudio: AAudioStream_requestStart(s#7) returned 0 ---------
2021-11-12 18:25:53.144 D/ViewRootImpl@be1f9c6[MainActivity]: ViewPostIme pointer 0
2021-11-12 18:25:53.183 D/ViewRootImpl@be1f9c6[MainActivity]: ViewPostIme pointer 1
2021-11-12 18:25:53.189 D/SystemWebChromeClient: : Line 506 : answer !
2021-11-12 18:25:53.189 I/chromium: [INFO:CONSOLE(506)] "answer !", source: (506)
2021-11-12 18:25:53.190 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SimpleUser | [Anonymous] Accepting Invitation...
2021-11-12 18:25:53.190 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SimpleUser | [Anonymous] Accepting Invitation...", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.190 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.Invitation | Invitation.accept
2021-11-12 18:25:53.190 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.Invitation | Invitation.accept", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.191 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.Invitation | Session 480a0aac-9522-448a-9b3a-0aa63a83df32ba5942c8-e85c-4993-a708-5a16c8442d9a transitioned to state Establishing
2021-11-12 18:25:53.191 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.Invitation | Session 480a0aac-9522-448a-9b3a-0aa63a83df32ba5942c8-e85c-4993-a708-5a16c8442d9a transitioned to state Establishing", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.191 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SimpleUser | [Anonymous] session state changed to Establishing
2021-11-12 18:25:53.191 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SimpleUser | [Anonymous] session state changed to Establishing", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.191 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | SessionDescriptionHandlerOptions: {}
2021-11-12 18:25:53.192 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | SessionDescriptionHandlerOptions: {}", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.192 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | initPeerConnection
2021-11-12 18:25:53.192 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | initPeerConnection", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.199 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | New peer connection created
2021-11-12 18:25:53.199 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | New peer connection created", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.209 D/InputMethodManager: HSIFW - flag : 0
2021-11-12 18:25:53.220 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | track added
2021-11-12 18:25:53.220 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | track added", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.221 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | acquiring local media
2021-11-12 18:25:53.221 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | acquiring local media", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.242 D/SystemWebChromeClient: onPermissionRequest: [android.webkit.resource.AUDIO_CAPTURE]
2021-11-12 18:25:53.503 13713-14416/com.weble.shs I/libOpenSLES: Emulating old channel mask behavior (ignoring positional mask 0x1, using default mask 0x10 based on channel count of 1)
2021-11-12 18:25:53.504 13713-14416/com.weble.shs D/AudioRecord: openRecord_l() remove fast or raw flag of voip calls, source=7, flags=5
2021-11-12 18:25:53.540 E/chromium: [ERROR:web_contents_delegate.cc(228)] WebContentsDelegate::CheckMediaAccessPermission: Not supported.
2021-11-12 18:25:53.540 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | acquired local media streams
2021-11-12 18:25:53.540 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | acquired local media streams", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.542 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | createAnswer
2021-11-12 18:25:53.542 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | createAnswer", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.544 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | resetIceGatheringComplete
2021-11-12 18:25:53.544 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | resetIceGatheringComplete", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.550 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | Setting local sdp.
2021-11-12 18:25:53.550 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | Setting local sdp.", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.551 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | sdp is v=0
o=- 8164219686908461201 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio-0
a=msid-semantic: WMS 3LWsKO9d6yj7LVeiLAXLUa8b6BVPakgBtqHy
m=audio 9 UDP/TLS/RTP/SAVPF 0 101
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:1kjN
a=ice-pwd:N8dI4ZN1IWcKkH9klCeFNFRF
a=ice-options:trickle
a=fingerprint:sha-256 7E:47:E8:64:4D:E8:55:7B:41:BE:0E:70:1E:FB:8C:F8:14:54:BA:F7:3E:5E:28:3C:D8:2B:38:68:3F:6F:D6:D5
a=setup:active
a=mid:audio-0
a=sendrecv
a=msid:3LWsKO9d6yj7LVeiLAXLUa8b6BVPakgBtqHy 67de7176-d33e-46ec-8983-c9e387e1b179
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ssrc:2672657289 cname:5Bn3TUwUFGZ1eI0d
2021-11-12 18:25:53.551 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | sdp is v=0
o=- 8164219686908461201 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio-0
a=msid-semantic: WMS 3LWsKO9d6yj7LVeiLAXLUa8b6BVPakgBtqHy
m=audio 9 UDP/TLS/RTP/SAVPF 0 101
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:1kjN
a=ice-pwd:N8dI4ZN1IWcKkH9klCeFNFRF
a=ice-options:trickle
a=fingerprint:sha-256 7E:47:E8:64:4D:E8:55:7B:41:BE:0E:70:1E:FB:8C:F8:14:54:BA:F7:3E:5E:28:3C:D8:2B:38:68:3F:6F:D6:D5
a=setup:active
a=mid:audio-0
a=sendrecv
a=msid:3LWsKO9d6yj7LVeiLAXLUa8b6BVPakgBtqHy 67de7176-d33e-46ec-8983-c9e387e1b179
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ssrc:2672657289 cname:5Bn3TUwUFGZ1eI0d
", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.557 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | waitForIceGatheringComplete
2021-11-12 18:25:53.557 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | waitForIceGatheringComplete", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.557 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | ICE is not complete. Returning promise
2021-11-12 18:25:53.557 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | ICE is not complete. Returning promise", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.558 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | RTCIceGatheringState changed: gathering
2021-11-12 18:25:53.558 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | RTCIceGatheringState changed: gathering", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.558 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | ICE Connection State changed to iceConnectionChecking
2021-11-12 18:25:53.558 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | ICE Connection State changed to iceConnectionChecking", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.559 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | ICE candidate received: candidate:947564071 1 udp 2113937151 10.83.8.129 35183 typ host generation 0 ufrag 1kjN network-cost 999
2021-11-12 18:25:53.559 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | ICE candidate received: candidate:947564071 1 udp 2113937151 10.83.8.129 35183 typ host generation 0 ufrag 1kjN network-cost 999", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.606 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | ICE candidate received: candidate:842163049 1 udp 1677729535 194.230.146.240 8547 typ srflx raddr 10.83.8.129 rport 35183 generation 0 ufrag 1kjN network-cost 999
2021-11-12 18:25:53.606 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | ICE candidate received: candidate:842163049 1 udp 1677729535 194.230.146.240 8547 typ srflx raddr 10.83.8.129 rport 35183 generation 0 ufrag 1kjN network-cost 999", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.656 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | RTCIceGatheringState changed: complete
2021-11-12 18:25:53.657 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | RTCIceGatheringState changed: complete", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.659 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.Transport | Sending WebSocket message:
SIP/2.0 200 OK
Via: SIP/2.0/WS 127.0.0.1:8088;rport;branch=z9hG4bKPja7de5ada-75f8-480e-9bfe-6d5d4bb5db75;alias
From: <sip:1001@XXXXXXXX>;tag=ba5942c8-e85c-4993-a708-5a16c8442d9a
To: <sip:[email protected]>;tag=02r8ufl7ki
CSeq: 16759 INVITE
Call-ID: 480a0aac-9522-448a-9b3a-0aa63a83df32
Supported: outbound
User-Agent: SIP.js/0.16.0
Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE
Contact: <sip:[email protected];transport=ws>
Content-Type: application/sdp
Content-Length: 893
v=0
o=- 8164219686908461201 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio-0
a=msid-semantic: WMS 3LWsKO9d6yj7LVeiLAXLUa8b6BVPakgBtqHy
m=audio 8547 UDP/TLS/RTP/SAVPF 0 101
c=IN IP4 194.230.146.240
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:947564071 1 udp 2113937151 10.83.8.129 35183 typ host generation 0 network-cost 999
a=candidate:842163049 1 udp 1677729535 194.230.146.240 8547 typ srflx raddr 10.83.8.129 rport 35183 generation 0 network-cost 999
a=ice-ufrag:1kjN
a=ice-pwd:N8dI4ZN1IWcKkH9klCeFNFRF
a=ice-options:trickle
a=fingerprint:sha-256 7E:47:E8:64:4D:E8:55:7B:41:BE:0E:70:1E:FB:8C:F8:14:54:BA:F7:3E:5E:28:3C:D8:2B:38:68:3F:6F:D6:D5
a=setup:active
a=mid:audio-0
a=sendrecv
a=msid:3LWsKO9d6yj7LVeiLAXLUa8b6BVPakgBtqHy 67de7176-d33e-46ec-8983-c9e387e1b179
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ssrc:2672657289 cname:5Bn3TUwUFGZ1eI0d
2021-11-12 18:25:53.659 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.Transport | Sending WebSocket message:
SIP/2.0 200 OK
Via: SIP/2.0/WS 127.0.0.1:8088;rport;branch=z9hG4bKPja7de5ada-75f8-480e-9bfe-6d5d4bb5db75;alias
From: <sip:1001@XXXXXXXX>;tag=ba5942c8-e85c-4993-a708-5a16c8442d9a
To: <sip:[email protected]>;tag=02r8ufl7ki
CSeq: 16759 INVITE
Call-ID: 480a0aac-9522-448a-9b3a-0aa63a83df32
Supported: outbound
User-Agent: SIP.js/0.16.0
Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE
Contact: <sip:[email protected];transport=ws>
Content-Type: application/sdp
Content-Length: 893
v=0
o=- 8164219686908461201 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio-0
a=msid-semantic: WMS 3LWsKO9d6yj7LVeiLAXLUa8b6BVPakgBtqHy
m=audio 8547 UDP/TLS/RTP/SAVPF 0 101
c=IN IP4 194.230.146.240
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:947564071 1 udp 2113937151 10.83.8.129 35183 typ host generation 0 network-cost 999
a=candidate:842163049 1 udp 1677729535 194.230.146.240 8547 typ srflx raddr 10.83.8.129 rport 35183 generation 0 network-cost 999
a=ice-ufrag:1kjN
a=ice-pwd:N8dI4ZN1IWcKkH9klCeFNFRF
a=ice-options:trickle
a=fingerprint:sha-256 7E:47:E8:64:4D:E8:55:7B:41:BE:0E:70:1E:FB:8C:F8:14:54:BA:F7:3E:5E:28:3C:D8:2B:38:68:3F:6F:D6:D5
a=setup:active
a=mid:audio-0
a=sendrecv
a=msid:3LWsKO9d6yj7LVeiLAXLUa8b6BVPakgBtqHy 67de7176-d33e-46ec-8983-c9e387e1b179
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ssrc:2672657289 cname:5Bn3TUwUFGZ1eI0d
", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.660 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.Invitation | Session 480a0aac-9522-448a-9b3a-0aa63a83df32ba5942c8-e85c-4993-a708-5a16c8442d9a transitioned to state Established
2021-11-12 18:25:53.660 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.Invitation | Session 480a0aac-9522-448a-9b3a-0aa63a83df32ba5942c8-e85c-4993-a708-5a16c8442d9a transitioned to state Established", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.660 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SimpleUser | [Anonymous] session state changed to Established
2021-11-12 18:25:53.660 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SimpleUser | [Anonymous] session state changed to Established", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.663 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | ICE candidate gathering complete
2021-11-12 18:25:53.664 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | ICE candidate gathering complete", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.674 E/chromium: [ERROR:web_contents_delegate.cc(228)] WebContentsDelegate::CheckMediaAccessPermission: Not supported.
2021-11-12 18:25:53.675 13713-14416/com.weble.shs I/AAudio: AAudioStreamBuilder_openStream() called ----------------------------------------
2021-11-12 18:25:53.675 13713-14416/com.weble.shs I/AudioStreamBuilder: rate = 48000, channels = 2, format = 5, sharing = SH, dir = OUTPUT
2021-11-12 18:25:53.675 13713-14416/com.weble.shs I/AudioStreamBuilder: device = 0, sessionId = -1, perfMode = 12, callback: ON with frames = 4104
2021-11-12 18:25:53.675 13713-14416/com.weble.shs I/AudioStreamBuilder: usage = 2, contentType = 0, inputPreset = 0, allowedCapturePolicy = 0
2021-11-12 18:25:53.675 13713-14416/com.weble.shs D/AudioStreamTrack: open(), request notificationFrames = -8, frameCount = 0
2021-11-12 18:25:53.684 13713-14416/com.weble.shs W/AudioTrack: createTrack_l(266): AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount 0 -> 4096
2021-11-12 18:25:53.684 13713-14416/com.weble.shs I/AudioTrack: Need throttle time for OpenSLES player
2021-11-12 18:25:53.684 13713-14416/com.weble.shs D/AudioTrack: setVolume(1.000000, 1.000000) pid : 13713
2021-11-12 18:25:53.685 13713-14416/com.weble.shs W/AudioStreamTrack: open() flags changed from 0x00000104 to 0x00000000
2021-11-12 18:25:53.685 13713-14416/com.weble.shs W/AudioStreamTrack: open() perfMode changed from 12 to 10
2021-11-12 18:25:53.686 13713-14416/com.weble.shs I/AAudio: AAudioStreamBuilder_openStream() returns 0 = AAUDIO_OK for s#8 ----------------
2021-11-12 18:25:53.687 13713-14416/com.weble.shs D/AAudio: AAudioStream_requestStart(s#8) called --------------
2021-11-12 18:25:53.689 13713-14416/com.weble.shs D/AAudio: AAudioStream_requestStart(s#8) returned 0 ---------
2021-11-12 18:25:53.704 13713-14365/com.weble.shs D/AudioStreamLegacy: onAudioDeviceUpdate() devId 3 => 3
2021-11-12 18:25:53.746 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.Transport | Received WebSocket text message:
ACK sip:[email protected]:44514;transport=ws SIP/2.0
Via: SIP/2.0/WS 127.0.0.1:8088;rport;branch=z9hG4bKPj9f428122-712e-4771-a8c8-bb6e37abda61;alias
From: <sip:1001@XXXXXXXX>;tag=ba5942c8-e85c-4993-a708-5a16c8442d9a
To: <sip:[email protected]>;tag=02r8ufl7ki
Call-ID: 480a0aac-9522-448a-9b3a-0aa63a83df32
CSeq: 16759 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-16-89cf7899be
Content-Length: 0
2021-11-12 18:25:53.746 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.Transport | Received WebSocket text message:
ACK sip:[email protected]:44514;transport=ws SIP/2.0
Via: SIP/2.0/WS 127.0.0.1:8088;rport;branch=z9hG4bKPj9f428122-712e-4771-a8c8-bb6e37abda61;alias
From: <sip:1001@XXXXXXXX>;tag=ba5942c8-e85c-4993-a708-5a16c8442d9a
To: <sip:[email protected]>;tag=02r8ufl7ki
Call-ID: 480a0aac-9522-448a-9b3a-0aa63a83df32
CSeq: 16759 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-16-89cf7899be
Content-Length: 0
", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.767 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.invite-dialog | INVITE dialog 480a0aac-9522-448a-9b3a-0aa63a83df3202r8ufl7kiba5942c8-e85c-4993-a708-5a16c8442d9a received ACK request
2021-11-12 18:25:53.767 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.invite-dialog | INVITE dialog 480a0aac-9522-448a-9b3a-0aa63a83df3202r8ufl7kiba5942c8-e85c-4993-a708-5a16c8442d9a received ACK request", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:53.768 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.Invitation | Session.onAckRequest
2021-11-12 18:25:53.768 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:53 GMT+0100 (heure normale d’Europe centrale) | sip.Invitation | Session.onAckRequest", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:55.366 D/ViewRootImpl@be1f9c6[MainActivity]: ViewPostIme pointer 0
2021-11-12 18:25:55.455 D/ViewRootImpl@be1f9c6[MainActivity]: ViewPostIme pointer 1
2021-11-12 18:25:55.460 D/SystemWebChromeClient: : Line 502 : hangup !
2021-11-12 18:25:55.460 I/chromium: [INFO:CONSOLE(502)] "hangup !", source: (502)
2021-11-12 18:25:55.460 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.SimpleUser | [Anonymous] Hangup...
2021-11-12 18:25:55.460 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.SimpleUser | [Anonymous] Hangup...", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:55.460 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.SimpleUser | [Anonymous] Terminating...
2021-11-12 18:25:55.460 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.SimpleUser | [Anonymous] Terminating...", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:55.461 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.invite-dialog | INVITE dialog 480a0aac-9522-448a-9b3a-0aa63a83df3202r8ufl7kiba5942c8-e85c-4993-a708-5a16c8442d9a sending BYE request
2021-11-12 18:25:55.461 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.invite-dialog | INVITE dialog 480a0aac-9522-448a-9b3a-0aa63a83df3202r8ufl7kiba5942c8-e85c-4993-a708-5a16c8442d9a sending BYE request", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:55.463 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.Transport | Sending WebSocket message:
BYE sip:asterisk@XXXXXXXX:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.0.2.243;branch=z9hG4bK1186987
To: <sip:1001@XXXXXXXX>;tag=ba5942c8-e85c-4993-a708-5a16c8442d9a
From: <sip:[email protected]>;tag=02r8ufl7ki
CSeq: 1 BYE
Call-ID: 480a0aac-9522-448a-9b3a-0aa63a83df32
Max-Forwards: 70
Supported: outbound
User-Agent: SIP.js/0.16.0
Content-Length: 0
2021-11-12 18:25:55.463 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.Transport | Sending WebSocket message:
BYE sip:asterisk@XXXXXXXX:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.0.2.243;branch=z9hG4bK1186987
To: <sip:1001@XXXXXXXX>;tag=ba5942c8-e85c-4993-a708-5a16c8442d9a
From: <sip:[email protected]>;tag=02r8ufl7ki
CSeq: 1 BYE
Call-ID: 480a0aac-9522-448a-9b3a-0aa63a83df32
Max-Forwards: 70
Supported: outbound
User-Agent: SIP.js/0.16.0
Content-Length: 0
", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:55.463 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.invite-dialog | INVITE dialog 480a0aac-9522-448a-9b3a-0aa63a83df3202r8ufl7kiba5942c8-e85c-4993-a708-5a16c8442d9a destroyed
2021-11-12 18:25:55.464 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.invite-dialog | INVITE dialog 480a0aac-9522-448a-9b3a-0aa63a83df3202r8ufl7kiba5942c8-e85c-4993-a708-5a16c8442d9a destroyed", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:55.464 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.Invitation | Session 480a0aac-9522-448a-9b3a-0aa63a83df32ba5942c8-e85c-4993-a708-5a16c8442d9a transitioned to state Terminated
2021-11-12 18:25:55.464 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.Invitation | Session 480a0aac-9522-448a-9b3a-0aa63a83df32ba5942c8-e85c-4993-a708-5a16c8442d9a transitioned to state Terminated", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:55.464 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.SimpleUser | [Anonymous] session state changed to Terminated
2021-11-12 18:25:55.464 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.SimpleUser | [Anonymous] session state changed to Terminated", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:55.465 13713-14416/com.weble.shs D/AAudio: AAudioStream_requestStop(s#8) called
2021-11-12 18:25:55.465 13713-14416/com.weble.shs D/AudioTrack: stop(271): called with 86912 frames delivered
2021-11-12 18:25:55.471 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.Invitation | Session 480a0aac-9522-448a-9b3a-0aa63a83df32ba5942c8-e85c-4993-a708-5a16c8442d9a in state Terminated is being disposed
2021-11-12 18:25:55.471 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.Invitation | Session 480a0aac-9522-448a-9b3a-0aa63a83df32ba5942c8-e85c-4993-a708-5a16c8442d9a in state Terminated is being disposed", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:55.471 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | closing PeerConnection
2021-11-12 18:25:55.471 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | closing PeerConnection", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:55.472 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | resetIceGatheringComplete
2021-11-12 18:25:55.472 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.SessionDescriptionHandler | resetIceGatheringComplete", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:55.480 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.SimpleUser | [Anonymous] Session ended (sent BYE)
2021-11-12 18:25:55.480 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.SimpleUser | [Anonymous] Session ended (sent BYE)", source: file:///android_asset/www/js/sip.js (1)
2021-11-12 18:25:55.496 D/InputMethodManager: HSIFW - flag : 0
2021-11-12 18:25:55.529 D/SystemWebChromeClient: file:///android_asset/www/js/sip.js: Line 1 : Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.Transport | Received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.243;rport=44514;received=10.11.72.99;branch=z9hG4bK1186987
Call-ID: 480a0aac-9522-448a-9b3a-0aa63a83df32
From: <sip:[email protected]>;tag=02r8ufl7ki
To: <sip:1001@XXXXXXXX>;tag=ba5942c8-e85c-4993-a708-5a16c8442d9a
CSeq: 1 BYE
Server: Asterisk PBX GIT-16-89cf7899be
Content-Length: 0
2021-11-12 18:25:55.529 I/chromium: [INFO:CONSOLE(1)] "Fri Nov 12 2021 18:25:55 GMT+0100 (heure normale d’Europe centrale) | sip.Transport | Received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.243;rport=44514;received=10.11.72.99;branch=z9hG4bK1186987
Call-ID: 480a0aac-9522-448a-9b3a-0aa63a83df32
From: <sip:[email protected]>;tag=02r8ufl7ki
To: <sip:1001@XXXXXXXX>;tag=ba5942c8-e85c-4993-a708-5a16c8442d9a
CSeq: 1 BYE
Server: Asterisk PBX GIT-16-89cf7899be
Content-Length: 0
", source: file:///android_asset/www/js/sip.js (1)
Maybe there is a link with this error :
2021-11-12 18:25:53.674 E/chromium: [ERROR:web_contents_delegate.cc(228)] WebContentsDelegate::CheckMediaAccessPermission: Not supported.
I'm able the answer the call, hangup it but no sound, no micro.
I hope you can help me ! Thanks for your help
Sipjs on commandline possible?
Hello,
Is it possible to use sip.js on the commandline? I would like to pipe video input from VLC/Ffmpeg and send it through sipjs.
I have seen your nodejs code examples and your website mentions that it can be used inside the browser or within Nodejs but when I use it in Nodejs, how can I pass the process.stdin as the video input to the library?
Thanks,
Marlo
Document Media Handler
The media handler is a pretty important part of SIP.js and it deserves some documentation.
Subscription's `failed` event is undocumented
While investigating onsip/SIP.js#247, I noticed that http://sipjs.com/api/0.7.0/subscription/ doesn't mention the failed event, which is emitted at: https://github.com/onsip/SIP.js/blob/0.7.1/src/Subscription.js#L263
Document `hold` and `unhold`
We're using proper re-INVITEs now, and so it is a good time to evaluate the browser landscape and see if hold and unhold are something that we can document into the actual public API.
SIP/2.0 401 Unauthorized with Asterisk
Hello, I have followed the official tutorial to configure Asterisk with sipjs but always I get the same error as "Unautheticated", here is the full log.
Wed Nov 03 2021 09:24:28 GMT+0100 (Central European Standard Time) | sip.SimpleUser | [Anonymous] Connecting UserAgent...
logger-factory.js:93 Wed Nov 03 2021 09:24:28 GMT+0100 (Central European Standard Time) | sip.UserAgent | Starting sip:[email protected]
logger-factory.js:93 Wed Nov 03 2021 09:24:28 GMT+0100 (Central European Standard Time) | sip.UserAgent | Transitioned from Stopped to Started
logger-factory.js:93 Wed Nov 03 2021 09:24:28 GMT+0100 (Central European Standard Time) | sip.Transport | Connecting ws://10.128.100.62:8088/ws
logger-factory.js:93 Wed Nov 03 2021 09:24:28 GMT+0100 (Central European Standard Time) | sip.Transport | Transitioned from Disconnected to Connecting
logger-factory.js:93 Wed Nov 03 2021 09:24:28 GMT+0100 (Central European Standard Time) | sip.Transport | WebSocket opened ws://10.128.100.62:8088/ws
logger-factory.js:93 Wed Nov 03 2021 09:24:28 GMT+0100 (Central European Standard Time) | sip.Transport | Transitioned from Connecting to Connected
logger-factory.js:93 Wed Nov 03 2021 09:24:28 GMT+0100 (Central European Standard Time) | sip.SimpleUser | [Anonymous] Connected
logger-factory.js:93 Wed Nov 03 2021 09:24:28 GMT+0100 (Central European Standard Time) | sip.SimpleUser | [Anonymous] Beginning Session...
logger-factory.js:93 Wed Nov 03 2021 09:24:28 GMT+0100 (Central European Standard Time) | sip.Inviter | Inviter.invite
logger-factory.js:93 Wed Nov 03 2021 09:24:33 GMT+0100 (Central European Standard Time) | sip.Inviter | Session d0i77ge4qmn6690nlu6c433s6ukn2q transitioned to state Establishing
logger-factory.js:93 Wed Nov 03 2021 09:24:33 GMT+0100 (Central European Standard Time) | sip.SimpleUser | [Anonymous] session state changed to Establishing
logger-factory.js:93 Wed Nov 03 2021 09:24:33 GMT+0100 (Central European Standard Time) | sip.Transport | Sending WebSocket message:INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WS qkuvnpcmvoau.invalid;branch=z9hG4bK2154278
To: sip:[email protected]
From: sip:[email protected];tag=433s6ukn2q
CSeq: 1 INVITE
Call-ID: d0i77ge4qmn6690nlu6c
Max-Forwards: 70
Contact: sip:[email protected];transport=ws;ob
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.20.0
Content-Type: application/sdp
Content-Length: 2459v=0
o=- 41774729151709639 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS abec084d-7ba7-4119-9b82-1650d96c4cb2
m=audio 50742 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 212.166.12.138
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1721360712 1 udp 2122260223 172.23.64.1 50740 typ host generation 0 network-id 2
a=candidate:1587246316 1 udp 2122194687 10.81.234.9 50741 typ host generation 0 network-id 3
a=candidate:404473800 1 udp 2122129151 10.176.36.10 50742 typ host generation 0 network-id 1 network-cost 10
a=candidate:3993746972 1 udp 1685921535 212.166.12.138 50742 typ srflx raddr 10.176.36.10 rport 50742 generation 0 network-id 1 network-cost 10
a=candidate:672800184 1 tcp 1518280447 172.23.64.1 9 typ host tcptype active generation 0 network-id 2
a=candidate:270043164 1 tcp 1518214911 10.81.234.9 9 typ host tcptype active generation 0 network-id 3
a=candidate:1452817208 1 tcp 1518149375 10.176.36.10 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:sygL
a=ice-pwd:5jLLqHIq1W76ujc6EliNPsNS
a=ice-options:trickle
a=fingerprint:sha-256 C1:17:34:F2:90:D9:9E:3F:F0:27:B0:F3:57:3D:D5:22:D8:CE:D0:6F:79:57:35:54:B3:B3:94:92:42:3C:34:4B
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:abec084d-7ba7-4119-9b82-1650d96c4cb2 d7105c38-691a-430d-9dbc-38f2b4067933
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:1550504661 cname:C/95kTYh2jyaLVa/
a=ssrc:1550504661 msid:abec084d-7ba7-4119-9b82-1650d96c4cb2 d7105c38-691a-430d-9dbc-38f2b4067933
a=ssrc:1550504661 mslabel:abec084d-7ba7-4119-9b82-1650d96c4cb2
a=ssrc:1550504661 label:d7105c38-691a-430d-9dbc-38f2b4067933logger-factory.js:93 Wed Nov 03 2021 09:24:33 GMT+0100 (Central European Standard Time) | sip.SimpleUser | [Anonymous] sent INVITE
logger-factory.js:93 Wed Nov 03 2021 09:24:33 GMT+0100 (Central European Standard Time) | sip.Transport | Received WebSocket text message:SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS qkuvnpcmvoau.invalid;rport=51388;received=10.81.234.9;branch=z9hG4bK2154278
Call-ID: d0i77ge4qmn6690nlu6c
From: sip:[email protected];tag=433s6ukn2q
To: sip:[email protected];tag=z9hG4bK2154278
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1635927930/6b5966f956de8a0b88cb7e6db335dd57",opaque="0bf928b66d3bc065",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.7.1
Content-Length: 0logger-factory.js:93 Wed Nov 03 2021 09:24:33 GMT+0100 (Central European Standard Time) | sip.Transport | Sending WebSocket message:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/WS qkuvnpcmvoau.invalid;branch=z9hG4bK2154278
To: sip:[email protected];tag=z9hG4bK2154278
From: sip:[email protected];tag=433s6ukn2q
Call-ID: d0i77ge4qmn6690nlu6c
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0logger-factory.js:93 Wed Nov 03 2021 09:24:33 GMT+0100 (Central European Standard Time) | sip.Transport | Sending WebSocket message:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WS qkuvnpcmvoau.invalid;branch=z9hG4bK6178544
To: sip:[email protected]
From: sip:[email protected];tag=433s6ukn2q
CSeq: 2 INVITE
Call-ID: d0i77ge4qmn6690nlu6c
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username="1060", realm="asterisk", nonce="1635927930/6b5966f956de8a0b88cb7e6db335dd57", uri="sip:[email protected]", response="f6f9e2d5720d22286fd9a87c464b7a43", opaque="0bf928b66d3bc065", qop=auth, cnonce="crir37rbf20a", nc=00000001
Contact: sip:[email protected];transport=ws;ob
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.20.0
Content-Type: application/sdp
Content-Length: 2459v=0
o=- 41774729151709639 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS abec084d-7ba7-4119-9b82-1650d96c4cb2
m=audio 50742 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 212.166.12.138
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1721360712 1 udp 2122260223 172.23.64.1 50740 typ host generation 0 network-id 2
a=candidate:1587246316 1 udp 2122194687 10.81.234.9 50741 typ host generation 0 network-id 3
a=candidate:404473800 1 udp 2122129151 10.176.36.10 50742 typ host generation 0 network-id 1 network-cost 10
a=candidate:3993746972 1 udp 1685921535 212.166.12.138 50742 typ srflx raddr 10.176.36.10 rport 50742 generation 0 network-id 1 network-cost 10
a=candidate:672800184 1 tcp 1518280447 172.23.64.1 9 typ host tcptype active generation 0 network-id 2
a=candidate:270043164 1 tcp 1518214911 10.81.234.9 9 typ host tcptype active generation 0 network-id 3
a=candidate:1452817208 1 tcp 1518149375 10.176.36.10 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:sygL
a=ice-pwd:5jLLqHIq1W76ujc6EliNPsNS
a=ice-options:trickle
a=fingerprint:sha-256 C1:17:34:F2:90:D9:9E:3F:F0:27:B0:F3:57:3D:D5:22:D8:CE:D0:6F:79:57:35:54:B3:B3:94:92:42:3C:34:4B
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:abec084d-7ba7-4119-9b82-1650d96c4cb2 d7105c38-691a-430d-9dbc-38f2b4067933
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:1550504661 cname:C/95kTYh2jyaLVa/
a=ssrc:1550504661 msid:abec084d-7ba7-4119-9b82-1650d96c4cb2 d7105c38-691a-430d-9dbc-38f2b4067933
a=ssrc:1550504661 mslabel:abec084d-7ba7-4119-9b82-1650d96c4cb2
a=ssrc:1550504661 label:d7105c38-691a-430d-9dbc-38f2b4067933logger-factory.js:93 Wed Nov 03 2021 09:24:33 GMT+0100 (Central European Standard Time) | sip.Transport | Received WebSocket text message:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS qkuvnpcmvoau.invalid;rport=51388;received=10.81.234.9;branch=z9hG4bK6178544
Call-ID: d0i77ge4qmn6690nlu6c
From: sip:[email protected];tag=433s6ukn2q
To: sip:[email protected];tag=z9hG4bK6178544
CSeq: 2 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1635927930/6b5966f956de8a0b88cb7e6db335dd57",opaque="144d99503710251b",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.7.1
Content-Length: 0logger-factory.js:93 Wed Nov 03 2021 09:24:33 GMT+0100 (Central European Standard Time) | sip.Transport | Sending WebSocket message:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/WS qkuvnpcmvoau.invalid;branch=z9hG4bK6178544
To: sip:[email protected];tag=z9hG4bK6178544
From: sip:[email protected];tag=433s6ukn2q
Call-ID: d0i77ge4qmn6690nlu6c
CSeq: 2 ACK
Max-Forwards: 70
Content-Length: 0
Respond to INFO requests in dialog
Currently, we only respond to DTMF INFO packets. We've had a request from FreeSWTICH to acknowledge the receipt of INFO packets. It shouldn't be hard to 200 Okay them.
Demo broken in Chrome 47
(originally noticed by @etamme)
While the demo on the front page worked in Chrome Version 46.0.2490.86 (64-bit), Chrome Version 47.0.2526.73 (64-bit) shows the following console error when a "video" button is clicked:
NavigatorUserMediaError {}
constraintName: ""
message: "Only secure origins are allowed (see: https://goo.gl/Y0ZkNV)."
name: "PermissionDeniedError"
__proto__: NavigatorUserMediaError
As https://goo.gl/Y0ZkNV vaguely suggests, Chrome 47 disables getUserMedia for untrusted origins. Since sipjs.com is currently hosted by Github Pages, we should be able to find more information here: isaacs/github#156
Single line code examples are difficult to read and understand.
I don't understand why, but the code examples on sipjs.com are in one line and difficult to read in browser. I use Chrome, latest build and have downloaded Firefox and even tried it on Safari, but the result is the same. This package is great but this makes this very difficult to learn and wrap your head around since it's not easy to work with example code blocks.
However, not all code blocks are like that. Rarely code is displayed correctly as the following screenshots demonstrate. This is all on the same page: https://sipjs.com/guides/simple/
It'd be great if all code appeared like this:
instead of this:
notice that I have to scroll the one liner to see the rest and the result isn't very pleasing nor useful:
This not only makes it very difficult to copy example code for active learning purposes, but it's not easy to read what's written as well.
SetTypingDetectionStatus: not supported
I've been trying to add sipjs 0.7.5 into a webview on Android but can't seem to get it to run correctly. I am able to get it to run on Chrome desktop browser in jsfiddle fine but in the webview it can't get a audio stream. Comparing the logs this is what I see as different.
09-16 22:45:56.873 11689-11689/com.app.dev I/chromium: [INFO:CONSOLE(2884)] "Fri Sep 16 2016 22:45:56 GMT-0700 (PDT) | sip.invitecontext.mediahandler | acquired local media streams", source: file:///android_asset/scripts/sip-0.7.5.js (2884) 09-16 22:45:56.883 11689-12572/com.app.dev W/chromium: [WARNING:mediasession.cc(328)] Duplicate id found. Reassigning from 101 to 127 09-16 22:45:56.895 11689-12573/com.app.dev E/chromium: [ERROR:voe_audio_processing_impl.cc(955)] SetTypingDetectionStatus: not supported 09-16 22:45:56.896 11689-12573/com.app.dev W/chromium: [WARNING:webrtcvoiceengine.cc(817)] SetTypingDetectionStatus(0) failed, err=8003 09-16 22:45:56.905 11689-12573/com.app.dev E/chromium: [ERROR:voe_audio_processing_impl.cc(955)] SetTypingDetectionStatus: not supported 09-16 22:45:56.905 11689-12573/com.app.dev W/chromium: [WARNING:webrtcvoiceengine.cc(817)] SetTypingDetectionStatus(0) failed, err=8003 09-16 22:45:56.911 11689-12573/com.app.dev W/chromium: [WARNING:stunport.cc(393)] Jingle:Port[0x848f1c00:audio:1:0::Net[any:0:0:0:x:x:x:x:x/0:Unknown]]: StunPort: stun host lookup received error 0 09-16 22:45:56.914 11689-12573/com.app.dev W/chromium: [WARNING:stunport.cc(393)] Jingle:Port[0x848f2100:audio:2:0::Net[any:0:0:0:x:x:x:x:x/0:Unknown]]: StunPort: stun host lookup received error 0
09-16 22:45:59.072 11689-12572/com.PrankRiot.dev E/chromium: [ERROR:webrtcsession.cc(1193)] SetAudioSend: No audio channel exists.
The code in the activity:
`
WebView webView = (WebView) findViewById(R.id.webview);
webView.setWebViewClient(new WebViewClient());
webView.setWebChromeClient(new WebChromeClient() {
@Override
public void onPermissionRequest(final PermissionRequest request) {
Log.d(TAG, "onPermissionRequest");
mActivity.runOnUiThread(new Runnable() {
@TargetApi(Build.VERSION_CODES.LOLLIPOP)
@Override
public void run() {
if(request.getOrigin().toString().equals("https://stream.demoapp.com/")) {
request.grant(request.getResources());
} else {
request.deny();
}
}
});
}
});
webView.addJavascriptInterface(new WebViewInterface(getApplicationContext()), "Android");
webView.getSettings().setJavaScriptEnabled(true);
webView.loadUrl("file:///android_asset/html/index.html");`
index.html:
`
The devices I've tried are Nexus 6(5.1.1) and Nexus 5x(7.0). Does the error "SetTypingDetectionStatus: not supported" matter? Comparing with desktop Chrome that seems to be the only difference and I'm running Asterisk and the SIP traffic looks normal till client requests to close the connection when the above errors happen.
Switch between LTE and WIFI
We use sip.js 0.7.5 for WebRTC on Cordova. It works fine with LTE and WiFi separately but after switching between LTE and WiFi, we lose the audio. We tried to close and unregister then reregister but we still get the same problem until we close the app and open it again. Is there a way to fix this issue? We also noticed that if I do that before first sip invite, we don't have the problem.
This is the log for the call on LTE. Destination answered and audio was fine.Then we switched to WiFi and called again.Destination answered but there was no audio.
broken
See onsip/SIP.js#930
Wrong "a=mid:" identification-tag in SDP for video
Hello dear SIP.js Team,
I have followiing behavior.
- I send an invite, which contain only audio:
v=0
o=- 6487371386765808549 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS b5fa3720-4d43-4908-83bf-922e3e85d189
m=audio 21218 UDP/TLS/RTP/SAVPF 111 63 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 195.70.114.111
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:36711831 1 tcp 1518283007 2a02:16a8:dc:aff:41ff:7a05:aabe:ea75 9 typ host tcptype active generation 0 network-id 4
a=candidate:1459951043 1 tcp 1518217471 2a02:16a8:dc:aff:bc63:1125:3d23:ccb7 9 typ host tcptype active generation 0 network-id 5
a=candidate:1127467076 1 tcp 1518149375 172.19.128.1 9 typ host tcptype active generation 0 network-id 1
a=candidate:940248979 1 tcp 1518083839 192.168.249.22 9 typ host tcptype active generation 0 network-id 2
a=candidate:696139521 1 tcp 1518018303 192.168.178.22 9 typ host tcptype active generation 0 network-id 3
a=candidate:1286852967 1 udp 2122262783 2a02:16a8:dc:aff:41ff:7a05:aabe:ea75 61929 typ host generation 0 network-id 4
a=candidate:428271923 1 udp 2122197247 2a02:16a8:dc:aff:bc63:1125:3d23:ccb7 61930 typ host generation 0 network-id 5
a=candidate:229651636 1 udp 2122129151 172.19.128.1 61931 typ host generation 0 network-id 1
a=candidate:1988686691 1 udp 2122063615 192.168.249.22 61932 typ host generation 0 network-id 2
a=candidate:1744684017 1 udp 2121998079 192.168.178.22 61933 typ host generation 0 network-id 3
a=candidate:2876308578 1 udp 1685790463 195.70.114.111 21218 typ srflx raddr 192.168.178.22 rport 61933 generation 0 network-id 3
a=candidate:3120942320 1 udp 1685855999 80.120.59.162 46995 typ srflx raddr 192.168.249.22 rport 61932 generation 0 network-id 2
a=ice-ufrag:XS3k
a=ice-pwd:j8j0QnwWORfxTD2uwXe2h4/L
a=ice-options:trickle
a=fingerprint:sha-256 BB:D7:1C:65:9F:65:4F:F1:BB:26:67:0F:B6:FD:51:D1:1F:42:54:65:3A:49:52:28:DD:38:4C:BF:A2:FD:CF:B4
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:b5fa3720-4d43-4908-83bf-922e3e85d189 6df34184-2a9d-47fe-85f0-96c2e4459064
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
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:173696517 cname:oOk4E6XnJI5qXBO4
a=ssrc:173696517 msid:b5fa3720-4d43-4908-83bf-922e3e85d189 6df34184-2a9d-47fe-85f0-96c2e4459064
a=ssrc:173696517 mslabel:b5fa3720-4d43-4908-83bf-922e3e85d189
a=ssrc:173696517 label:6df34184-2a9d-47fe-85f0-96c2e4459064
- I get a Re-Invite back from other Peer, which contains audio and video:
v=0
o=- 2587202469 5 IN IP4 168.63.9.134
s=Commend
c=IN IP4 168.63.9.134
t=0 0
m=audio 27736 RTP/SAVP 111 9 8 0
a=maxptime:20
a=mid:0
a=rtpmap:111 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=fmtp:111 useinbandfec=1
a=sendrecv
a=rtcp:27737
a=rtcp-mux
a=setup:actpass
a=fingerprint:sha-1 72:AD:6C:9B:65:58:60:32:28:6C:69:9F:0C:A7:B1:36:E7:67:1F:8E
a=ptime:20
a=ice-ufrag:KgNA0ytg
a=ice-pwd:9auF5FrqwACfGoX3zIQeXX3LtZ
a=ice-options:trickle
a=candidate:sItLLPWORbggr2b1 1 UDP 2130706431 168.63.9.134 27736 typ host
a=candidate:sItLLPWORbggr2b1 2 UDP 2130706430 168.63.9.134 27737 typ host
a=end-of-candidates
m=video 27804 RTP/SAVP 99
a=rtpmap:99 H264/90000
a=sendrecv
a=rtcp:27805
a=rtcp-mux
a=setup:actpass
a=fingerprint:sha-1 72:AD:6C:9B:65:58:60:32:28:6C:69:9F:0C:A7:B1:36:E7:67:1F:8E
a=ice-ufrag:NxZM89gT
a=ice-pwd:0hBE3a6zujN7gJBonZhxrHwVVK
a=candidate:sItLLPWORbggr2b1 1 UDP 2130706431 168.63.9.134 27804 typ host
a=candidate:sItLLPWORbggr2b1 2 UDP 2130706430 168.63.9.134 27805 typ host
As you can see there is no a=mid attribute for video.
Question: Can the missing mid attribute (in the Re-Invite) lead to a=mid:2 attribute for video in a 200 Ok Response?
Because this is the outcome in this SDP:
v=0
o=- 6487371386765808549 3 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS b5fa3720-4d43-4908-83bf-922e3e85d189
m=audio 21218 RTP/SAVP 111 9 8 0
c=IN IP4 195.70.114.111
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:36711831 1 tcp 1518283007 2a02:16a8:dc:aff:41ff:7a05:aabe:ea75 9 typ host tcptype active generation 0 network-id 4
a=candidate:1459951043 1 tcp 1518217471 2a02:16a8:dc:aff:bc63:1125:3d23:ccb7 9 typ host tcptype active generation 0 network-id 5
a=candidate:1127467076 1 tcp 1518149375 172.19.128.1 9 typ host tcptype active generation 0 network-id 1
a=candidate:940248979 1 tcp 1518083839 192.168.249.22 9 typ host tcptype active generation 0 network-id 2
a=candidate:696139521 1 tcp 1518018303 192.168.178.22 9 typ host tcptype active generation 0 network-id 3
a=candidate:1286852967 1 udp 2122262783 2a02:16a8:dc:aff:41ff:7a05:aabe:ea75 61929 typ host generation 0 network-id 4
a=candidate:428271923 1 udp 2122197247 2a02:16a8:dc:aff:bc63:1125:3d23:ccb7 61930 typ host generation 0 network-id 5
a=candidate:229651636 1 udp 2122129151 172.19.128.1 61931 typ host generation 0 network-id 1
a=candidate:1988686691 1 udp 2122063615 192.168.249.22 61932 typ host generation 0 network-id 2
a=candidate:1744684017 1 udp 2121998079 192.168.178.22 61933 typ host generation 0 network-id 3
a=candidate:2876308578 1 udp 1685790463 195.70.114.111 21218 typ srflx raddr 192.168.178.22 rport 61933 generation 0 network-id 3
a=candidate:3120942320 1 udp 1685855999 80.120.59.162 46995 typ srflx raddr 192.168.249.22 rport 61932 generation 0 network-id 2
a=ice-ufrag:XS3k
a=ice-pwd:j8j0QnwWORfxTD2uwXe2h4/L
a=ice-options:trickle
a=fingerprint:sha-256 BB:D7:1C:65:9F:65:4F:F1:BB:26:67:0F:B6:FD:51:D1:1F:42:54:65:3A:49:52:28:DD:38:4C:BF:A2:FD:CF:B4
a=setup:passive
a=mid:0
a=sendrecv
a=msid:b5fa3720-4d43-4908-83bf-922e3e85d189 6df34184-2a9d-47fe-85f0-96c2e4459064
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ssrc:173696517 cname:oOk4E6XnJI5qXBO4
a=ssrc:173696517 msid:b5fa3720-4d43-4908-83bf-922e3e85d189 6df34184-2a9d-47fe-85f0-96c2e4459064
a=ssrc:173696517 mslabel:b5fa3720-4d43-4908-83bf-922e3e85d189
a=ssrc:173696517 label:6df34184-2a9d-47fe-85f0-96c2e4459064
m=video 64355 RTP/SAVP 99
c=IN IP4 172.19.128.1
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:36711831 1 tcp 1518283007 2a02:16a8:dc:aff:41ff:7a05:aabe:ea75 9 typ host tcptype active generation 0 network-id 4
a=candidate:1459951043 1 tcp 1518217471 2a02:16a8:dc:aff:bc63:1125:3d23:ccb7 9 typ host tcptype active generation 0 network-id 5
a=candidate:1127467076 1 tcp 1518149375 172.19.128.1 9 typ host tcptype active generation 0 network-id 1
a=candidate:940248979 1 tcp 1518083839 192.168.249.22 9 typ host tcptype active generation 0 network-id 2
a=candidate:696139521 1 tcp 1518018303 192.168.178.22 9 typ host tcptype active generation 0 network-id 3
a=candidate:1286852967 1 udp 2122262783 2a02:16a8:dc:aff:41ff:7a05:aabe:ea75 64353 typ host generation 0 network-id 4
a=candidate:428271923 1 udp 2122197247 2a02:16a8:dc:aff:bc63:1125:3d23:ccb7 64354 typ host generation 0 network-id 5
a=candidate:229651636 1 udp 2122129151 172.19.128.1 64355 typ host generation 0 network-id 1
a=candidate:1988686691 1 udp 2122063615 192.168.249.22 64356 typ host generation 0 network-id 2
a=candidate:1744684017 1 udp 2121998079 192.168.178.22 64357 typ host generation 0 network-id 3
a=ice-ufrag:c25J
a=ice-pwd:xXnYN0tY48GjKzldtGnXM5Y2
a=ice-options:trickle
a=fingerprint:sha-256 BB:D7:1C:65:9F:65:4F:F1:BB:26:67:0F:B6:FD:51:D1:1F:42:54:65:3A:49:52:28:DD:38:4C:BF:A2:FD:CF:B4
a=setup:active
a=mid:2
a=sendrecv
a=msid:- 53e27296-b5bd-4d5a-b0e0-8707094a7086
a=rtcp-mux
a=rtpmap:99 H264/90000
a=fmtp:99 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
a=ssrc:1216605974 cname:oOk4E6XnJI5qXBO4
a=ssrc:1216605974 msid:- 53e27296-b5bd-4d5a-b0e0-8707094a7086
a=ssrc:1216605974 mslabel:-
a=ssrc:1216605974 label:53e27296-b5bd-4d5a-b0e0-8707094a7086
Video contains a=mid:2, although it should be a=mid:1
How can this be?
Version: 0.20.0
Thank you very much!
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