unable to get remote stream in case when a “remote stream” is attached by calling “peer.addStream(remoteStream) about webrtc-experiment

[gatherhub]

Hi Muaz, I read through this thread and run your demo it seems that adding remote stream to another peerconnection does work. Howerver, when I wrote my own code, I cached the remote stream and added it to another peerconnection, the far end does not get onaddstream event fired and cannot get the forwarded stream. There was no errors and iceconnectionstate is completed too. I have also checked the ice candidates information. There are all generated completely and correctly without duplication to the original peerconnection. Would you help to provide some hint? Thank you very much!

[atsepkov]

@muaz-khan For solving the problem you describe with peer-5 leaving, could the server instead tell peer 6 or 7 (who should be in the same link in the chain as 5) to take over instead? Similarly, could this system be made redundant/safe by having each child peer be aware of all 3 parent peers in the chain and fallback automatically if one drops?

[HiroakiLion]

Any progress on this module?
I'm looking for a solution for 100-200 listeners vs 1 streamer.

[muaz-khan]

unable to generate ice candidates (on the answerer side) when attaching remote stream..

a=candidate:2437072876 1 udp 2113937151 192.168.1.2 54082 typ host generation 0
a=candidate:2437072876 2 udp 2113937151 192.168.1.2 54082 typ host generation 0
a=candidate:2437072876 1 udp 2113937151 192.168.1.2 54082 typ host generation 0
a=candidate:2437072876 2 udp 2113937151 192.168.1.2 54082 typ host generation 0
a=candidate:941443129 1 udp 1845501695 39.47.69.177 10136 typ srflx raddr 192.168.1.2 rport 54082 generation 0
a=candidate:941443129 2 udp 1845501695 39.47.69.177 10136 typ srflx raddr 192.168.1.2 rport 54082 generation 0
a=candidate:941443129 1 udp 1845501695 39.47.69.177 10136 typ srflx raddr 192.168.1.2 rport 54082 generation 0
a=candidate:941443129 2 udp 1845501695 39.47.69.177 10136 typ srflx raddr 192.168.1.2 rport 54082 generation 0

--------offer sdp provided by offerer

v=0
o=- 2216700829 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS
m=audio 1 RTP/SAVPF 103 104 111 0 8 107 106 105 13 126
c=IN IP4 0.0.0.0
a=rtcp:1 IN IP4 0.0.0.0
a=ice-ufrag:Yy0jsw+XcU7Ovmw4
a=ice-pwd:9kjBUxNEoOOO0yViUNEH9Yhe
a=ice-options:google-ice
a=fingerprint:sha-256 ED:B9:CF:19:5F:A3:F4:D3:31:FC:F5:18:3C:AF:60:9E:B1:52:A5:32:69:8A:22:63:05:AD:98:93:ED:33:2A:7E
a=recvonly
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:CFkD+WaozWTFg41l4mi5rdoAscTYMJa8CB1HgVpW
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 CN/48000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
m=video 1 RTP/SAVPF 100 116 117
c=IN IP4 0.0.0.0
a=rtcp:1 IN IP4 0.0.0.0
a=ice-ufrag:Yy0jsw+XcU7Ovmw4
a=ice-pwd:9kjBUxNEoOOO0yViUNEH9Yhe
a=ice-options:google-ice
a=fingerprint:sha-256 ED:B9:CF:19:5F:A3:F4:D3:31:FC:F5:18:3C:AF:60:9E:B1:52:A5:32:69:8A:22:63:05:AD:98:93:ED:33:2A:7E
a=recvonly
a=mid:video
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:CFkD+WaozWTFg41l4mi5rdoAscTYMJa8CB1HgVpW
a=rtpmap:100 VP8/90000
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
RTCPeerConnection-v1.4.js:191

--------answer

sdp: v=0
o=- 315182876 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS
m=audio 1 RTP/SAVPF 103 104 111 0 8 107 106 105 13 126
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Do0n1o1bjqhwgfo7jsfqBi2pgBx9i9egkjngrjkf
c=IN IP4 0.0.0.0
a=rtcp:1 IN IP4 0.0.0.0
a=ice-ufrag:ZQglAeIL9/0MtR+R
a=ice-pwd:gbAu8knZaPhZrBQc9mDjK2AT
a=fingerprint:sha-256 50:DD:32:5B:31:92:C0:CF:77:2A:DC:66:45:29:6C:58:73:A8:77:AF:10:38:D0:FA:53:D9:55:26:0A:DC:B2:6A
a=recvonly
a=mid:audio
a=rtcp-mux
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 CN/48000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
m=video 1 RTP/SAVPF 100 116 117
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Do0n1o1bjqhwgfo7jsfqBi2pgBx9i9egkjngrjkf
c=IN IP4 0.0.0.0
a=rtcp:1 IN IP4 0.0.0.0
a=ice-ufrag:ZQglAeIL9/0MtR+R
a=ice-pwd:gbAu8knZaPhZrBQc9mDjK2AT
a=fingerprint:sha-256 50:DD:32:5B:31:92:C0:CF:77:2A:DC:66:45:29:6C:58:73:A8:77:AF:10:38:D0:FA:53:D9:55:26:0A:DC:B2:6A
a=recvonly
a=mid:video
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000

Maybe RTP ports already opened....We may need to change them dynamically.

[muaz-khan]

Latest one-page demo on attaching remote media streams. Still fails.

[peili]

Any process on this issue so far?

[muaz-khan]

This issue is not fixed yet. Attachment of remote media streams still failing on both chrome and firefox. I tried audio-only remote streams too but no success.

[peili]

Do you know when Google/Mozilla are going to support this feature? I couldn't find clear informations on the roadmaps..

[muaz-khan]

I'm not sure, however it seems that audio-only remote streams' attachment works fine.

[muaz-khan]

A general comment is that we only support one audio source today in Chrome given that we only have one ADM. To me it sounds like we need a new type of source here where the audio is driven by the remote peer. Perhaps it would be less complicated to build something initially if we only worked in relay mode (one source).

Reference: https://code.google.com/p/webrtc/issues/detail?id=383

.....take a remote stream from one peer connection and adding it to another peer connection and having it Just Work™ for that second peer.

Implementation awaited!

[RonMen]

Hi @muaz-khan,

I think this is what I thought about on my other issue questioning. The use case is:

1.) Peer1 connects with audio only
2.) Peer2 connects with audio only
... later on one/both switch
3.) Peer1/Peer2 switches over to remove audio only stream and instead add a video+audio stream

In this case I get a black video on the peer this stream is delivered to sending a "new" offer and there are removeStream + addStream events fired up but also the data of the newly added stream isn't send.

I tried multiple settings and also tried to find an answer but it doesn't seem to work. Also the sample implementation presented in the webrtc issue tracker isn't working like expected: https://code.google.com/p/webrtc/issues/detail?id=1086

Do you have any idea how to bring this up and running or is there something missing in browsers webrtc implementation for now?

[muaz-khan]

Try this demo: Switching streams from screen-sharing to audio+video at runtime. (Renegotiation)

1. Both peers are sharing screen
2. You can switch to audio+video stream; old stream (i.e. screen-sharing) will be removed.

Don't need to add ice candidates when renegotiating.

[RonMen]

Dear @muaz-khan, thx for your sample. I also found something else since setting the localDescription failed while switching like explained here: https://groups.google.com/forum/#!msg/discuss-webrtc/ets-eex6H-0/bkN2UwZI-gEJ

I changed the {optional: [{DtlsSrtpKeyAgreement: true}]} of the peerconnection and I tried without it and the streams are switched well now.

[skanagavelu]

We need this idea. This communication is low cost for end user.
Mauz; How this is supported for screen share?

[skanagavelu]

Hi Muaz,

In your example below

firstPeer.onaddstream = function(remoteSteam) {
remoteStream = new MediaStream(remoteSteam.audioTracks, remoteSteam.videoTracks);
otherPeer.addStream(remoteStream); /* attaching remote stream */
};

Why we are recreating remoteStream again using new MediaStream ?

And pls tell me what we have to do for screen share while creating new instance of MediaStream...?

[muaz-khan]

There are some bugs preventing video tracks to be forwarded. In screen sharing apps; we need to retransmit video RTP packets; which seems failing on both chrome and firefox.

In the example; new media stream is created; because in old days, chrome were throwing not-supported exceptions for MediaStream because only LocalMediaStream objects were permitted to be attached using peer.addStream method.

Nowadays, you can attach remote stream directly, on chrome; without initiating new object; because both local and remote streams have identical MediaStream type.

I think identical ports (RTP or UDP) are causing failures to attach remote media streams. Though, I don't know the exact issue. Maybe ICE Agent is unable to locate right ports to retransmit remote media stream.

Maximum peer connections limit is 256; so it is preferred to use application specific bandwidth parameters to stream low-quality packets; also specify width/height constraints to handle freezing issues.

[skanagavelu]

FYI : https://groups.google.com/forum/#!topic/discuss-webrtc/CnEHFxjxE_8

[peili]

Any updates on "attaching remote audio/video stream"?
Audio seems to work. Video not?

[skanagavelu]

https://code.google.com/p/webrtc/issues/detail?can=2&start=0&num=100&q=&colspec=ID%20Pri%20Mstone%20ReleaseBlock%20Area%20Status%20Owner%20Summary&groupby=&sort=&id=383

[skanagavelu]

https://code.google.com/p/webrtc/issues/detail?id=2192

[abhishanksahu]

Hi Muaz, Is there any progress in this issue? I am able to attach remote streaming on the same page. But when I try to do the same over two different pages it is failing.

[muaz-khan]

It works. Try following demo on chrome stable (desktop):

<script src="//www.webrtc-experiment.com/RTCMultiConnection-v1.6.js"> </script>

<button id="open-main-session">open main session</button>
<button id="forward-main-session">forward main session</button><br />

<button id="join-main-session">join main session</button>
<button id="join-forwarded-session">join forwarded session</button>
<script>
var h1 = document.createElement('h1');
h1.innerHTML = 'user-id: ' + prompt('What is Your Name');
document.body.insertBefore(h1, document.body.firstChild);

// http://www.rtcmulticonnection.org/docs/constructor/
var mainConnection = new RTCMultiConnection();

// http://www.rtcmulticonnection.org/docs/session/
mainConnection.session = {
    video: true,
    oneway: true
};

// http://www.rtcmulticonnection.org/docs/constructor/
var dummyConnection = new RTCMultiConnection('dummy-connection');

// http://www.rtcmulticonnection.org/docs/session/
dummyConnection.session = {
    video: true,
    oneway: true
};

// http://www.rtcmulticonnection.org/docs/dontAttachStream/
dummyConnection.dontAttachStream = true;

// http://www.rtcmulticonnection.org/docs/onstream/
mainConnection.onstream = function(e) {
    // see this line--------------------------
    dummyConnection.attachStreams.push(e.stream);
    document.body.appendChild(e.mediaElement);
};

// http://www.rtcmulticonnection.org/docs/onstream/
dummyConnection.onstream = function(e) {
    alert('Wow, got forwarded stream!');
    document.body.appendChild(e.mediaElement);
};

document.querySelector('#join-main-session').onclick = function() {
    this.disabled = true;

    // http://www.rtcmulticonnection.org/docs/connect/
    mainConnection.connect();
};

document.querySelector('#join-forwarded-session').onclick = function() {
    this.disabled = true;

    // http://www.rtcmulticonnection.org/docs/connect/
    dummyConnection.connect();
};

document.querySelector('#open-main-session').onclick = function() {
    this.disabled = true;

    // http://www.rtcmulticonnection.org/docs/open/
    mainConnection.open();
};

document.querySelector('#forward-main-session').onclick = function() {
    this.disabled = true;

    // http://www.rtcmulticonnection.org/docs/open/
    dummyConnection.open();
};
</script>

dummyConnection is used to forward media stream. Main forwarding stuff happened here:

mainConnection.onstream = function(e) {
    dummyConnection.attachStreams.push(e.stream);
};

[abhishanksahu]

Hi Muaz,

Thanks for your help, I tried this demo and its work fine for video, but
when I enable audio also, on the forwarded stream audio is missing, there
is no sound from the third tab. Can you tell me why?

Thank You

[muaz-khan]

Did you appended audio:true in the session object?

mainConnection.session = {
    audio: true,
    video: true,
    oneway: true
};

[abhishanksahu]

Yeah I did that, on the second tab I am able to listen audio, but when on
third tab I join the forwarded session, I can only see the video and there
is no sound coming from it. With latest version of chrome u can see which
tab is playing audio right? There was no such sign over this tab.

On 5 February 2014 22:52, Muaz Khan [email protected] wrote:

Did you appended audio:true in the sessionhttp://www.rtcmulticonnection.org/docs/session/object?

mainConnection.session = {
audio: true,
video: true,
oneway: true};

Reply to this email directly or view it on GitHubhttps://github.com//issues/2#issuecomment-34213645
.

[harshg0910]

@muaz-khan
@abhishanksahu
I did some experiments with audio only :
Computer 1 :

1. Opened the main session in one tab
2. Joined the main session and forwarded it in the other tab of same computer.
   Computer 2
3. Joined the forwarded session but the audio player was there with no sound.
4. Joined the main session in other tab and I was able to hear the sound.
5. Now the tab containing the forwarded stream was also playing the sound ( I checked it by pausing the sound of the main session tab).
6. I close the tab with main session, the sound from the forwarded session also dies.

Seems interesting as I am unable to find the reason for failure of forwarded stream at the same time its relation with main stream.

[pandalaw]

Hi Muaz,

I tried to forward stream of screen, but it failed. Can you tell me why?
Computer 2 cannot get the screen of computer 1 as well.

Thank You

[muaz-khan]

I tried this demo; changed this line:

mainConnection.session = {
    screen: true,
    oneway: true
};

It worked when I shared screen from canary. In the below gif; you can see that I tested between chrome canary, chrome stable and opera next 19.0.

[abhishanksahu]

Hi Muaz,

Did you find any remedy for the problem I described you (no sound from the
forwarded stream).

Thanks.

On 5 February 2014 22:56, Abhishank Sahu [email protected] wrote:

Yeah I did that, on the second tab I am able to listen audio, but when on
third tab I join the forwarded session, I can only see the video and there
is no sound coming from it. With latest version of chrome u can see which
tab is playing audio right? There was no such sign over this tab.

On 5 February 2014 22:52, Muaz Khan [email protected] wrote:

Did you appended audio:true in the sessionhttp://www.rtcmulticonnection.org/docs/session/object?

mainConnection.session = {
audio: true,
video: true,
oneway: true};

Reply to this email directly or view it on GitHubhttps://github.com//issues/2#issuecomment-34213645
.

[muaz-khan]

See this comment:

Due to some architecture problems in the chromium remote audio track implementation, today we are always mixing the data in WebRtc, and passing the mixed data to Chrome. This ways prevents us getting the correct stream for the specific audio track, and it also blocks us feeding the correct remote audio track data to the peer connection.

After talking to a couple of colleagues about this feature, we decided to postpone this feature until we fix the architecture problems.

That's why remote audio recording doesn't work.

[pandalaw]

Hi Muaz,

I tried this demo. and changed line 135

mainConnection.session = {
    video: true,
    audio: true,
    screen: true,
    oneway: true
};

When I joined forwarded session, I can get screen stream only.

Please help me to find out the problem :)

Thank you.

[muaz-khan]

After looking into logs; it is appeared that only single "remote" media stream is attached to peer connection object:

In theory, it MUST attach both audio+video and screen streams. You can see that we are trying to attach both:

After going a little bit deeper; it is appeared that both remote streams has identical ids and labels. It can be RTCMultiConnection implementation issue.

[shtefcs]

Hi Muaz,

Is this problem fixed by chrome guys ?

[josephfrazier]

@shtefcs, it's been quite a while since you asked, but yes, Chrome 49 (Canary, at the time of writing) should:

• Implement adding remote audio track/stream to a PeerConnection
• Hook up Web Audio API with WebRTC for audio processing

[spozitor]

Any peer output without a "closed" sign, causes the remaining peers seek to reconnect?
You can 100,000 users?

[maurodelazeri]

what happens with the pear 60 when pear 59 was disconnected ?

[muaz-khan]

If peer-59 leaves, peer-60 is told to rejoin-broadcast. This is exactly how RTCMultiConnection is currently supporting it.

If peer-60 will rejoin, server will give him fresh relaying-peer-id. During this rejoin-process, RTCMultiConnection tries to play recorded stream until fresh stream-event is fired.

If peer-5 leaves then all-kinds-of nested child are told to rejoin. Currently this mechanism hangs the browser. Better solution will be only top nested child should be asked to rejoin; however other deeper nested child should renegotiate connections to get fresh stream(s).

[visionsTeam]

@muaz-khan
Hello,
I have successfully implemented your code on my php website and its very useful really appreciative. But the only issue i am facing is that when one of the user is on chrome browser and other in mozilla
when they share their webcam the video freeze and one of the user automatically left the room.
it will be very kind if you suggest some.

[naeemaziz]

Dear Muaz,

hope you are doing great. I need your help, I have install asterisk on Ubuntu and now with nodeserver am trying to connect sip with the websocket using webrtc. I have followed thhe link below to connect asterisk.
https://webrtc.ventures/2017/02/kurento-asterisk-powerful-couple/

The error which i am getting on the webrtc console is mention below.

Senging message: {"id":"register","ext":"2000","password":"asteriskpass"}
create webRtcPeer ...
spec: {"audio":true,"video":{"width":640,"framerate":15}}
chrome: {"audio":true,"video":{"optional":[{"minWidth":640},{"maxWidth":640},{"minFramerate":15},{"maxFramerate":15}]}}
created webRtcPeer
Generate offer
Senging message: {"id":"call","to":"2001","sdpOffer":"v=0\r\no=- 5684911275500021011 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video\r\na=msid-semantic: WMS 1zjjtovdbXvu9vpNwHvR5EuTFhZJ5ow9V4SY\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:SyyF\r\na=ice-pwd:6MMfMUAhfTmyP2rOv+TyXgEx\r\na=ice-options:trickle\r\na=fingerprint:sha-256 84:98:D4:4B:05:2C:08:D8:2A:A0:BB:4E:39:CD:F9:A4:C6:5C:22:84:D2:C0:D2:50:CD:2E:82:21:C1:C2:EF:AB\r\na=setup:actpass\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=recvonly\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:110 telephone-event/48000\r\na=rtpmap:112 telephone-event/32000\r\na=rtpmap:113 telephone-event/16000\r\na=rtpmap:126 telephone-event/8000\r\nm=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 123 127 122 125 107 108 109 124\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:SyyF\r\na=ice-pwd:6MMfMUAhfTmyP2rOv+TyXgEx\r\na=ice-options:trickle\r\na=fingerprint:sha-256 84:98:D4:4B:05:2C:08:D8:2A:A0:BB:4E:39:CD:F9:A4:C6:5C:22:84:D2:C0:D2:50:CD:2E:82:21:C1:C2:EF:AB\r\na=setup:actpass\r\na=mid:video\r\na=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:4 urn:3gpp:video-orientation\r\na=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=sendrecv\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:96 VP8/90000\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtpmap:97 rtx/90000\r\na=fmtp:97 apt=96\r\na=rtpmap:98 VP9/90000\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=rtpmap:99 rtx/90000\r\na=fmtp:99 apt=98\r\na=rtpmap:100 H264/90000\r\na=rtcp-fb:100 goog-remb\r\na=rtcp-fb:100 transport-cc\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f\r\na=rtpmap:101 rtx/90000\r\na=fmtp:101 apt=100\r\na=rtpmap:102 H264/90000\r\na=rtcp-fb:102 goog-remb\r\na=rtcp-fb:102 transport-cc\r\na=rtcp-fb:102 ccm fir\r\na=rtcp-fb:102 nack\r\na=rtcp-fb:102 nack pli\r\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=rtpmap:123 rtx/90000\r\na=fmtp:123 apt=102\r\na=rtpmap:127 H264/90000\r\na=rtcp-fb:127 goog-remb\r\na=rtcp-fb:127 transport-cc\r\na=rtcp-fb:127 ccm fir\r\na=rtcp-fb:127 nack\r\na=rtcp-fb:127 nack pli\r\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032\r\na=rtpmap:122 rtx/90000\r\na=fmtp:122 apt=127\r\na=rtpmap:125 H264/90000\r\na=rtcp-fb:125 goog-remb\r\na=rtcp-fb:125 transport-cc\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 nack\r\na=rtcp-fb:125 nack pli\r\na=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032\r\na=rtpmap:107 rtx/90000\r\na=fmtp:107 apt=125\r\na=rtpmap:108 red/90000\r\na=rtpmap:109 rtx/90000\r\na=fmtp:109 apt=108\r\na=rtpmap:124 ulpfec/90000\r\na=ssrc-group:FID 1296757455 1269725181\r\na=ssrc:1296757455 cname:SQMZcE5d/QTvKSfl\r\na=ssrc:1296757455 msid:1zjjtovdbXvu9vpNwHvR5EuTFhZJ5ow9V4SY a2e6dc91-0c6e-4ce4-9ae5-9d8ddae6d06c\r\na=ssrc:1296757455 mslabel:1zjjtovdbXvu9vpNwHvR5EuTFhZJ5ow9V4SY\r\na=ssrc:1296757455 label:a2e6dc91-0c6e-4ce4-9ae5-9d8ddae6d06c\r\na=ssrc:1269725181 cname:SQMZcE5d/QTvKSfl\r\na=ssrc:1269725181 msid:1zjjtovdbXvu9vpNwHvR5EuTFhZJ5ow9V4SY a2e6dc91-0c6e-4ce4-9ae5-9d8ddae6d06c\r\na=ssrc:1269725181 mslabel:1zjjtovdbXvu9vpNwHvR5EuTFhZJ5ow9V4SY\r\na=ssrc:1269725181 label:a2e6dc91-0c6e-4ce4-9ae5-9d8ddae6d06c\r\n"}
Local candidate{"candidate":"candidate:2172407820 1 udp 2122260223 10.10.2.128 53774 typ host generation 0 ufrag SyyF network-id 1 network-cost 10","sdpMid":"audio","sdpMLineIndex":0}
Senging message: {"id":"onIceCandidate","candidate":{"candidate":"candidate:2172407820 1 udp 2122260223 10.10.2.128 53774 typ host generation 0 ufrag SyyF network-id 1 network-cost 10","sdpMid":"audio","sdpMLineIndex":0}}
Local candidate{"candidate":"candidate:2172407820 1 udp 2122260223 10.10.2.128 53775 typ host generation 0 ufrag SyyF network-id 1 network-cost 10","sdpMid":"video","sdpMLineIndex":1}
Senging message: {"id":"onIceCandidate","candidate":{"candidate":"candidate:2172407820 1 udp 2122260223 10.10.2.128 53775 typ host generation 0 ufrag SyyF network-id 1 network-cost 10","sdpMid":"video","sdpMLineIndex":1}}
Local candidate{"candidate":"candidate:3489639676 1 tcp 1518280447 10.10.2.128 9 typ host tcptype active generation 0 ufrag SyyF network-id 1 network-cost 10","sdpMid":"audio","sdpMLineIndex":0}
Senging message: {"id":"onIceCandidate","candidate":{"candidate":"candidate:3489639676 1 tcp 1518280447 10.10.2.128 9 typ host tcptype active generation 0 ufrag SyyF network-id 1 network-cost 10","sdpMid":"audio","sdpMLineIndex":0}}
Local candidate{"candidate":"candidate:3489639676 1 tcp 1518280447 10.10.2.128 9 typ host tcptype active generation 0 ufrag SyyF network-id 1 network-cost 10","sdpMid":"video","sdpMLineIndex":1}
Senging message: {"id":"onIceCandidate","candidate":{"candidate":"candidate:3489639676 1 tcp 1518280447 10.10.2.128 9 typ host tcptype active generation 0 ufrag SyyF network-id 1 network-cost 10","sdpMid":"video","sdpMLineIndex":1}}
Communication ended by remote peer

and the error i am getting in CLi is mention below.

2000 calling 2001 ...
TERMINATED Not Found
TERMINATED Not Found
Tue Jun 05 2018 12:37:02 GMT+0500 (PKT) | sip.inviteclientcontext | closing INVITE session btgk632evrb42viujmfd8smg3kcme7
TERMINATED Not Found
Tue Jun 05 2018 12:37:02 GMT+0500 (PKT) | sip.transaction.ict | Timer D expired for INVITE client transaction z9hG4bK5247184
Connection 1 received message { id: 'onIceCandidate',
candidate:
{ candidate: 'candidate:2172407820 1 udp 2122260223 10.10.2.128 53774 typ host generation 0 ufrag SyyF network-id 1 network-cost 10',
sdpMid: 'audio',
sdpMLineIndex: 0 } }
Connection 1 received message { id: 'onIceCandidate',
candidate:
{ candidate: 'candidate:2172407820 1 udp 2122260223 10.10.2.128 53775 typ host generation 0 ufrag SyyF network-id 1 network-cost 10',
sdpMid: 'video',
sdpMLineIndex: 1 } }
Connection 1 received message { id: 'onIceCandidate',
candidate:
{ candidate: 'candidate:3489639676 1 tcp 1518280447 10.10.2.128 9 typ host tcptype active generation 0 ufrag SyyF network-id 1 network-cost 10',
sdpMid: 'audio',
sdpMLineIndex: 0 } }
Connection 1 received message { id: 'onIceCandidate',
candidate:
{ candidate: 'candidate:3489639676 1 tcp 1518280447 10.10.2.128 9 typ host tcptype active generation 0 ufrag SyyF network-id 1 network-cost 10',
sdpMid: 'video',
sdpMLineIndex: 1 } }
reconnect to server 0 100 undefined
reconnect to server 1 100 undefined
reconnect to server 2 200 undefined
reconnect to server 3 300 undefined
reconnect to server 4 500 undefined
reconnect to server 5 800 undefined
reconnect to server 6 1300 undefined
reconnect to server 7 2100 undefined
reconnect to server 8 3400 undefined
reconnect to server 9 5500 undefined
reconnect to server 10 8900 undefined
reconnect to server 11 10000 undefined
reconnect to server 12 10000 undefined
reconnect to server 13 10000 undefined
reconnect to server 14 10000 undefined
reconnect to server 15 10000 undefined
reconnect to server 16 10000 undefined
reconnect to server 17 10000 undefined
reconnect to server 18 10000 undefined
reconnect to server 19 10000 undefined
reconnect to server 20 10000 undefined
reconnect to server 21 10000 undefined
reconnect to server 22 10000 undefined
reconnect to server 23 10000 undefined
reconnect to server 24 10000 undefined
reconnect to server 25 10000 undefined
reconnect to server 26 10000 undefined
reconnect to server 27 10000 undefined
reconnect to server 28 10000 undefined
reconnect to server 29 10000 undefined
reconnect to server 30 10000 undefined
reconnect to server 31 10000 undefined
reconnect to server 32 10000 undefined
reconnect to server 33 10000 undefined
reconnect to server 34 10000 undefined
reconnect to server 35 10000 undefined
reconnect to server 36 10000 undefined
reconnect to server 37 10000 undefined
reconnect to server 38 10000 undefined
reconnect to server 39 10000 undefined
reconnect to server 40 10000 undefined
reconnect to server 41 10000 undefined

Can you lease help me out to resolve this issue?

[ankur2194]

@naeemaziz the reconnect log you received means the node.js application is not able to connect your kurento server.

[babycommando]

Hey, how does it work for around +1.000 users?