codec Transcoding

Linux Teki

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Sep 11, 2019, 1:26:19 AM9/11/19

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Hi All,

I have a janus setup using sip plugin to make sip-webrtc audio call and I have a scenario, where my browser uses only opus codec and my asterisk using only ilbc codec. So how can we make an audio to work in this scenrio? As I read from google we can do this by codec trans-coding(codec conversion) but I am not sure where and how I can implement this.

Can someone help me out on this please?

Regards,

Aravind

Mirko Brankovic

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Sep 11, 2019, 2:07:25 AM9/11/19

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By libc you are referring to AAC?

I'm sure Astsrisk is supporting Opus and high likely able to transcode from Opus to AAC

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Linux Teki

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Sep 11, 2019, 3:09:12 AM9/11/19

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Hi Mirko,

I am not sure what AAC is, however i am using iLBC audio codec on my asterisk server. Also, i didn't find AAC codec in my asterisk server.

I'm sure Astsrisk is supporting Opus and high likely able to transcode from Opus to AAC?

yeah, the latest version of asterisk is supporting opus, but i am using a bit old version of asterisk(1.2v) in which there was no opus support. Although there is g711 codec on both ends, due to bandwidth constraint i'm not able to use that. so i'm opted for iLBC.

Regards,

Arvind

Mirko Brankovic

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Sep 11, 2019, 4:34:28 AM9/11/19

to Linux Teki, meetecho-janus

If you are worried about bandwidth then it is best to use Opus.

So when comparing:

https://en.wikipedia.org/wiki/Comparison_of_audio_coding_formats

iLBC - Fixed bitrate (15.2 kbit/s for 20 ms frames, 13.33 kbit/s for 30 ms frames)

G729 - Fixed bit rate (8 kbit/s 10 ms frames)

Opus supports constant and variable bitrate encoding from 6 kbit/s to 510 kbit/s

I don't see G729 in SDP offer from Chrome, so not sure if it supports it at all, but asterisk and freeswitch surely can,

But opus can go even twice lower than iLBC in narrowband mode.

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Regards,

Mirko

Lorenzo Miniero

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Sep 11, 2019, 4:43:17 AM9/11/19

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We don't do transcoding in the SIP plugin and never will. Either add Opus to your super-old Asterisk version (we have an old patch for that: https://github.com/meetecho/asterisk-opus ) or have the Opus/iLBC transcoding happen somewhere else.

Lorenzo

Il giorno mercoledì 11 settembre 2019 10:34:28 UTC+2, Mirko Brankovic ha scritto:

If you are worried about bandwidth then it is best to use Opus.
So when comparing:
https://en.wikipedia.org/wiki/Comparison_of_audio_coding_formats
iLBC - Fixed bitrate (15.2 kbit/s for 20 ms frames, 13.33 kbit/s for 30 ms frames)
G729 - Fixed bit rate (8 kbit/s 10 ms frames)
Opus supports constant and variable bitrate encoding from 6 kbit/s to 510 kbit/s

I don't see G729 in SDP offer from Chrome, so not sure if it supports it at all, but asterisk and freeswitch surely can,
But opus can go even twice lower than iLBC in narrowband mode.

On Wed, Sep 11, 2019 at 9:09 AM Linux Teki <aravin...@gmail.com> wrote:

Hi Mirko,

I am not sure what AAC is, however i am using iLBC audio codec on my asterisk server. Also, i didn't find AAC codec in my asterisk server.

I'm sure Astsrisk is supporting Opus and high likely able to transcode from Opus to AAC?

yeah, the latest version of asterisk is supporting opus, but i am using a bit old version of asterisk(1.2v) in which there was no opus support. Although there is g711 codec on both ends, due to bandwidth constraint i'm not able to use that. so i'm opted for iLBC.

Regards,
Arvind

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Regards,
Mirko

Alexandre GOUAILLARD

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Sep 11, 2019, 5:02:37 AM9/11/19

to Lorenzo Miniero, meetecho-janus

G729 is not supported by chrome and they publicly said they would not support it (even though it is now public domain).

On Wed, Sep 11, 2019 at 10:43 AM Lorenzo Miniero <lmin...@gmail.com> wrote:

We don't do transcoding in the SIP plugin and never will. Either add Opus to your super-old Asterisk version (we have an old patch for that: https://github.com/meetecho/asterisk-opus ) or have the Opus/iLBC transcoding happen somewhere else.

Lorenzo

Il giorno mercoledì 11 settembre 2019 10:34:28 UTC+2, Mirko Brankovic ha scritto:

If you are worried about bandwidth then it is best to use Opus.
So when comparing:
https://en.wikipedia.org/wiki/Comparison_of_audio_coding_formats
iLBC - Fixed bitrate (15.2 kbit/s for 20 ms frames, 13.33 kbit/s for 30 ms frames)
G729 - Fixed bit rate (8 kbit/s 10 ms frames)
Opus supports constant and variable bitrate encoding from 6 kbit/s to 510 kbit/s

I don't see G729 in SDP offer from Chrome, so not sure if it supports it at all, but asterisk and freeswitch surely can,
But opus can go even twice lower than iLBC in narrowband mode.

On Wed, Sep 11, 2019 at 9:09 AM Linux Teki <aravin...@gmail.com> wrote:

Hi Mirko,

I am not sure what AAC is, however i am using iLBC audio codec on my asterisk server. Also, i didn't find AAC codec in my asterisk server.

I'm sure Astsrisk is supporting Opus and high likely able to transcode from Opus to AAC?

yeah, the latest version of asterisk is supporting opus, but i am using a bit old version of asterisk(1.2v) in which there was no opus support. Although there is g711 codec on both ends, due to bandwidth constraint i'm not able to use that. so i'm opted for iLBC.

Regards,
Arvind

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Regards,
Mirko

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Mirko Brankovic

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Sep 11, 2019, 8:51:20 AM9/11/19

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Yeah, silicone valley wars :D

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Regards,

Mirko

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