ILBC codec support

[Linux Teki]

Hi ALL,

I would like know if iLBC codec is enabled on webrtc or not. As per few articles, it was enabled on webrtc but not on the chrome browser. Is this true?

[Linux Teki]

Can someone help me on this please?

[Philipp Hancke]

paste this into the JS console:

  const pc = new RTCPeerConnection();

  pc.createOffer({offerToReceiveAudio: true}).then(offer => console.log(offer.sdp))

If ilbc shows up in the SDP its supported. It is not.

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[Linux Teki]

Hi Philipp,

Thanks for your response. Seems it didn't support ILBC as I can't find in SDP. So now I am thinking to go for transcoding from opus to ILBC as my sip endpoint supports only Ilbc codec. But as a newbie I have few questions can u clarify them.

1) Is it possible to transcode opus to ILBC.(Although g711 supports on both sides, due to bandwidth constraint we are not able to use that so opted for codec transcoding)

2) if yes, how&where to implement it? As i am not much of aware of this can you provide some example code? 

Below is my setup.

BROWSER----> JANUS----->Asterisk----->SIP END point

On Monday, December 16, 2019 at 7:38:57 PM UTC+5:30, Linux Teki wrote:

[Tsahi Levent-Levi]

Hi,

You should be able to transcode iLBC to Opus in Asterisk itself (or FreeSwitch - whichever you fancy).

There is another possibility for your architecture which is to forgo the use of Janus by just using Asterisk in the middle:

BROWSER---->Asterisk----->SIP END point 

To do that, you can use JsSIP in the browser and connect to Asterisk using SIP over WebSocket. You'll still need to transcode in Asterisk the voice codec though.

 

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Regards,

Tsahi Levent-Levi

Analyst & Consultant

BlogGeek.me

Want to get more of your WebRTC sessions effectivly connected? Enroll to my free video course: http://bit.ly/32Y3qZK

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Regards,

Tsahi Levent-Levi

Analyst & Consultant

BlogGeek.me

Want to get more of your WebRTC sessions effectivly connected? Enroll to my free video course: http://bit.ly/32Y3qZK

[Linux Teki]

Hi Tsahi,

Thanks for your quick reply. Yes, we are aware that asterisk can work as webrtc server and transcoder as well, however, we are using a bit older version of asterisk(v1.2) which was customized and stabilized. So as far as I know we don't have webrtc capabilities in that version and we not gonna update it as of now due to stability concern(may be in future we can), so we opted for janus to do the things.  

 Now the issue is our asterisk supports ILBC/Speex apart from g711, and webrtc don't have these codecs. So we are looking to have a transcoding from opus to ILBC.

Can you help me out on this please, as i'm new to transcoding where and how todo?

On Monday, December 16, 2019 at 7:38:57 PM UTC+5:30, Linux Teki wrote:

[Tsahi Levent-Levi]

Teki,

You'll need to either Opus codec to your Asterisk 12 or upgrade it...

The other alternative is to add another Asterisk of a newer version in front of the one you have and have that deal with transcoding. If I were you, I'd rather put the effort in upgrading it and be done with it (especially since you're running an old and unsupported version) - I'd go for a 16.x release.

Tsahi

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