Browser-Phone
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InnovateAsterisk
Speaker issue
There are a number of reason, you will have to explain further - could be as simple as not having interacting with the page yet.
This issue occurs when connected through mobile data(4G network). In the company's wifi its working fine.
Some carriers block voip traffic, or firewall the connection. You can try request a voip APN, but first make sure that STUN is setup (On the Asterisk side) working correctly.
https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/
| 0.002 | rtp | host | 1204296370 | udp | b0085051-30aa-4002-9f17-eab24e2acd62.local | 55271 | 126 | 30 | 255 | 0 | 0 | VhTn |
|---|---|---|---|---|---|---|---|---|---|---|
| 0.518 | rtp | srflx | 842163049 | udp | 111.92.32.12 | 55271 | 100 | 30 | 255 | 0 | 0 | VhTn |
| 39.827 | Done |
| Time | Component | Type | Foundation | Protocol | Address | Port | Priority | Mid | MLine Index | Username Fragment |
|---|---|---|---|---|---|---|---|---|---|---|
| 0.023 | rtp | host | 0 | udp | 4b10b754-9d4b-415f-83ea-50950913d681.local | 57316 | 126 | 32512 | 255 | 0 | 0 | 3cd2b7ef |
| 0.024 | rtp | host | 2 | tcp | 4b10b754-9d4b-415f-83ea-50950913d681.local | 9 | 125 | 32704 | 255 | 0 | 0 | 3cd2b7ef |
| 0.024 | rtcp | host | 0 | udp | 4b10b754-9d4b-415f-83ea-50950913d681.local | 57317 | 126 | 32512 | 254 | 0 | 0 | 3cd2b7ef |
| 0.024 | rtcp | host | 2 | tcp | 4b10b754-9d4b-415f-83ea-50950913d681.local | 9 | 125 | 32704 | 254 | 0 | 0 | 3cd2b7ef |
| 0.115 | rtp | srflx | 1 | udp | 111.92.32.21 | 57316 | 100 | 32543 | 255 | 0 | 0 | 3cd2b7ef |
| 0.153 | rtcp | srflx | 1 | udp | 111.92.32.21 | 57317 | 100 | 32543 | 254 | 0 | 0 | 3cd2b7ef |
| 0.155 | Done |
[Intervention] Slow network is detected. See https://www.chromestatus.com/feature/5636954674692096 for more details. Fallback font will be used while loading: https://dtd6jl0d42sve.cloudfront.net/lib/fonts/font_roboto/d-6IYplOFocCacKzxwXSOFtXRa8TVwTICgirnJhmVJw.woff2
Unable to play audio file. DOMException: play() failed because the user didn't interact with the document first.
Some carriers block voip traffic, or firewall the connection. You can try request a voip APN, but first make sure that STUN is setup (On the Asterisk side) working correctly.
But whatsapp audio and video calls are working fine. Google meet is also working.
srflx | udp | 111.92.32.12
This is a good result, you are able to determine the IP address from within your browser.
Also make sure that Asterisk is able to do the same (ICE setting must be applied to rtp.conf). I would assume that your Asterisk box is hosted via VPC or something. This NAT arrangement must be known to Asterisk, so that the Media address can be advertised.
Will this happen due to slow network?
https://stackoverflow.com/questions/40143098/why-does-this-slow-network-detected-log-appear-in-chrome
Unable to play audio file. DOMException: play() failed because the user didn't interact with the document first.
It is required that a user clicks or interacts with the page before the <audio> elements are able to play out. For example clicking "answer" will activate the audio. (however the inbound ring tone may not be able to play without a user clicking on the page). (Both iOS and Android have a flag to disable this requirement)
But whatsapp audio and video calls are working fine. Google meet is also working.
They probably don't use SIP.
which audio and video encoding are used in Browser phone?
It depends on the browser, and you should be able to see each type in the offer - Typically, Chrome will offer ULAW, ALAW and OPUS for audio and for video it it will be VP8 and VP9. (I know that Firefox prefers MP4)
Also make sure that Asterisk is able to do the same (ICE setting must be applied to rtp.conf). I would assume that your Asterisk box is hosted via VPC or something. This NAT arrangement must be known to Asterisk, so that the Media address can be advertised.
ICE Setting is included in rtp.conf How to check this?
stunaddr=stun.l.google.com:19302
stunaddr=stun.l.google.com:19302
This should be fine, but also check the nat settings especially this:
https://github.com/asterisk/asterisk/blob/892c06564f667ea438815afdac9ace929a346bab/configs/samples/rtp.conf.sample#L131
I'm getting this issue in firefox:
WebRTC: ICE failed, add a TURN server and see about:webrtc for more details
So, it was working, now its not?
What changed?
Do you have a console log of the "peer connection undefined" error. This is typically an issue if the offer fails to apply.
audio will probably not work if the peer connection is undefined.
So, it was working, now its not?
What changed?
Do you have a console log of the "peer connection undefined" error. This is typically an issue if the offer fails to apply.
audio will probably not work if the peer connection is undefined.
When I tested with vi network it worked. When tested with BSNL network this issue happens.
How is connection getting established even without the STUN/TURN server?