InnovateAsterisk / Browser-Phone

A fully featured browser based WebRTC SIP phone for Asterisk
https://www.innovateasterisk.com
GNU Affero General Public License v3.0
499 stars 245 forks source link

not getting video #439

Open prathibhacdac opened 1 year ago

prathibhacdac commented 1 year ago

video works in the same network but not in different network.

prathibhacdac commented 1 year ago

stun server issue is resolved. but not getting audio and video.

InnovateAsterisk commented 1 year ago

video works in the same network but not in different network.

Is this still the case?

prathibhacdac commented 1 year ago

Now I'm getting video in different n/w. But before sometime, I didn't get it.

InnovateAsterisk commented 1 year ago

Whats the latency (roundtrip) between your client and the server?

prathibhacdac commented 1 year ago

How to find latency?

Sent from Outlook for Androidhttps://aka.ms/AAb9ysg


From: Conrad de Wet @.> Sent: Tuesday, February 28, 2023 5:48:46 PM To: InnovateAsterisk/Browser-Phone @.> Cc: prathibhacdac @.>; Author @.> Subject: Re: [InnovateAsterisk/Browser-Phone] not getting video (Issue #439)

Whats the latency (roundtrip) between your client and the server?

— Reply to this email directly, view it on GitHubhttps://github.com/InnovateAsterisk/Browser-Phone/issues/439#issuecomment-1448086246, or unsubscribehttps://github.com/notifications/unsubscribe-auth/AGJHM3STVO3GQCIPBPU3P4TWZXUKNANCNFSM6AAAAAAU4RM7P4. You are receiving this because you authored the thread.Message ID: @.***! com>

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prathibhacdac commented 1 year ago

Traceroute from mobile to server: No. of hops 12, time 32094ms

Traceroute from PC to server: traceroute 10.10.10.73 traceroute to 10.10.10.73 (10.10.10.73), 30 hops max, 60 byte packets 1 bp1.erss.in (10.10.10.73) 0.039 ms 0.007 ms 0.010 ms

prathibhacdac commented 1 year ago

Today, I'm not getting the audio and video.

prathibhacdac commented 1 year ago

Is it required to open port 5060 in the asterisk server?

InnovateAsterisk commented 1 year ago

Is it required to open port 5060 in the asterisk server?

WebRTC does not use port 5060.

InnovateAsterisk commented 1 year ago

How to find latency?

For the moment a simple ping would give you an indication of how long messages take.

prathibhacdac commented 1 year ago

From mobile: Total execution time 12097ms

from PC Approximate round trip times in milli-seconds: Minimum = 6ms, Maximum = 7ms, Average = 6ms

InnovateAsterisk commented 1 year ago

Total execution time 12097ms

It’s pretty much impossible to do any kind of voice communication with a latency of 12sconds.

Minimum = 6ms, Maximum = 7ms, Average = 6ms

This is better, but now almost seems too quick. Is this local lan ?

prathibhacdac commented 1 year ago

Minimum = 6ms, Maximum = 7ms, Average = 6ms

This is better, but now almost seems too quick. Is this local lan ? Yes.

InnovateAsterisk commented 1 year ago

You need to perform some tests to find out at what point the video is failing. To do this, you should eliminate some of the call legs. Create a dial-plan entry to perform a simple echo with video. https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_StreamEcho Something like this should work. The calls should then be answer by asterisk and simply echoed back to you.

Configure something like the above, and test with your pc, and with the mobile. Note that you will not be calling between the devices. The cal flow will be from the client to the server and back only.

prathibhacdac commented 1 year ago

I'm able to hear the audio back from PC. In mobile, I'm getting video but it is struck.

InnovateAsterisk commented 1 year ago

If you connect the mobile phone to the same network as the pc using WiFi, what’s are the results?

prathibhacdac commented 1 year ago

Only getting Trying... on both sides.

prathibhacdac commented 1 year ago

STUN transaction has timed out (PJNATH_ESTUNTIMEDOUT). Is this an issue?