video is not getting display

[prathibhatvm]

In video call, only the video of the creator is displayed. The joiner's video is not appearing.

[InnovateAsterisk]

You'll have to give me a bit more to go with here... My tests show video working fine.

Can you send the Browser console Log and SIP trace

[prathibhacdac]

console log attached console.log

[prathibhacdac]

sipTrace.txt

[InnovateAsterisk]

Thu Jan 06 2022 09:28:52 GMT+0530 (India Standard Time) | sip.ua | · userAgentString: "Raspberry Phone (SipJS - 0.11.6)"

This is a log from an older version of the browser phone. This is known to have issues with video calls especially in conference (SFU) mode.

[prathibhacdac]

console1.log

[InnovateAsterisk]

My log appears almost the same as yours except that I get the following:

Playing Video Stream MID: 1 Display Video - live MID: 1 channel: unknown src width: 477 src height 270

What's interesting is that your log show:

phone.js:2466 Adding Remote Video Track - live MID: 1

and also:

Playing Video Stream MID: 1

But this was after the BYE in both cases.

This means that the streams are coming through to the browser because of the Playing Video, but because they are after the bye, it doesn't display them Display Video.

It's almost as if it's just taking a very long time. Is it possible that the video has just taken too long to stream? in some of my tests the video stream has taken a very long time to appear. I did notice in the previous log, the ICE collection timed out. You need to fix that - ICE is important for webrtc. Make sure you can reach "stun:stun.l.google.com:19302"

[prathibhacdac]

How to check stun:stun.l.google.com:19302 is reachable or not?

[InnovateAsterisk]

https://github.com/InnovateAsterisk/Browser-Phone/issues/182#issuecomment-1002936027

[prathibhacdac]

trickleice report

[InnovateAsterisk]

Looks good. You should not have any issues with ICE from client side.

Remember tho, ICE must be enabled on the server too. because with Asterisk Server is B-side always, even when calling another webrtc extension. (it cannot optimise out, as SDES is already established.)

https://github.com/asterisk/asterisk/blob/0d62735f99b27a7104bbdb04bb5630fec395f40f/configs/samples/rtp.conf.sample#L61

and also here if NAT: https://github.com/asterisk/asterisk/blob/0d62735f99b27a7104bbdb04bb5630fec395f40f/configs/samples/rtp.conf.sample#L131

[prathibhacdac]

Now I'm able to see both the creator and the joiner's video. Issue resolved.

[InnovateAsterisk]

Excellent!

BTW, If you dial into a confbridge conference https://github.com/asterisk/asterisk/blob/master/configs/samples/confbridge.conf.sample and set video_mode = sfu https://github.com/asterisk/asterisk/blob/0d62735f99b27a7104bbdb04bb5630fec395f40f/configs/samples/confbridge.conf.sample#L222 You can have a multi party video conference.

[prathibhacdac]

How to do video conference using Browser Phone?