VP8 Codec

Hello, are there plans to include VP8 or VP9 into distribution?

Last I knew Asterisk did not include the VP8 codec for video at this time. The distro will support the codecs that Asterisk itself has support for.

It’s in Asterisk 13:

asterisk*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.

  30 image      png (PNG Image)
   5 audio     g726 (G.726 RFC3551)
   3 audio     alaw (G.711 a-law)
   1 audio     g723 (G.723.1)
  19 audio    speex (SpeeX)
  20 audio    speex (SpeeX 16khz)
  21 audio    speex (SpeeX 32khz)
  23 audio     g722 (G722)
  31 video     h261 (H.261 video)
  32 video     h263 (H.263 video)
   7 audio    adpcm (Dialogic ADPCM)
  24 audio   siren7 (ITU G.722.1 (Siren7, licensed from Polycom))
  27 audio     g719 (ITU G.719)
  33 video    h263p (H.263+ video)
  34 video     h264 (H.264 video)
  18 audio     g729 (G.729A)
   8 audio     slin (16 bit Signed Linear PCM)
   9 audio     slin (16 bit Signed Linear PCM (12kHz))
  10 audio     slin (16 bit Signed Linear PCM (16kHz))
  11 audio     slin (16 bit Signed Linear PCM (24kHz))
  12 audio     slin (16 bit Signed Linear PCM (32kHz))
  13 audio     slin (16 bit Signed Linear PCM (44kHz))
  14 audio     slin (16 bit Signed Linear PCM (48kHz))
  15 audio     slin (16 bit Signed Linear PCM (96kHz))
  16 audio     slin (16 bit Signed Linear PCM (192kHz))
   2 audio     ulaw (G.711 u-law)
  17 audio    lpc10 (LPC10)
  26 audio  testlaw (G.711 test-law)
  39 audio     none (<Null> codec)
  25 audio  siren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))
   6 audio g726aal2 (G.726 AAL2)
  36 video      vp8 (VP8 video)
   4 audio      gsm (GSM)
  35 video    mpeg4 (MPEG4 video)
  22 audio     ilbc (iLBC)
  37  text      red (T.140 Realtime Text with redundancy)
  38  text     t140 (Passthrough T.140 Realtime Text)
  28 audio     opus (Opus Codec)
  29 image     jpeg (JPEG image)

That’s pass-through support for VP8 only, not transcoding of VP8.

Ok - When will we get Transcoding?

When someone takes on the rather large project of doing such a thing.

What would you do with it if you had it?

Well…having just got back from Astricon and being INCREDIBLY PUMPED about almost everything I saw and all the people I talked too, I don’t know, but I am itching to try out some WebRTC anything - wasn’t excitement about where Asterisk is going the whole point of Astricon?

Spcifically to your question “What would you do with it if you had it?” I will give the stock Field Of Dreams answer: “Build it and they will come.”

Unless I am a VERY bad futurist, I see WebRTC as the next really HUGE thing! And I want Asterisk to be in the center of the action, not on the side lines.

Is there a coding project for the codec already underway that I could help with?

Ok - So after I posted this, I had to run my daughter to Kung Fu and my mind wandered and now here is where I think I could use WebRTC and where it could go from there (about 45 minutes of pondering)

First: How about a click-to-call link on your web-page that instead of initiating a Voice Call, right from the web, it initiated a WebRTC call that hit your Asterisk box and was routed to your WebRTC-Enabled Receptionist that routed the Call just like any other? And even if where the call ended up was not WebRTC compliant, that Asterisk bridged the call so that whoever ended up with the call was bridged to the caller in at least a voice-usable format if they couldn’t negotiate WebRTC, or perhaps a Video Phone and Asterisk bridged the SIP Video to WebRTC and then back again.

And then I got thinking about a program like iSymphony with a Contact List that had all of the possible ways to contact a person and a rules-based system so that it could try them in the order you preferred so WebRTC->SIP Video-> SIP Audio-> PSTN-2-Cel -> SMS -> E-Mail

Eventually, I see Asterisk as being the Communications HUB for any business, being able to basically accept any offered medium of communication and using the same call-flow logic that we have now, but expanding it to intelligently handle all sorts of media,

Imagine a Social Network that would allow any member to talk with any other member, but also for a premium (or perhaps a freemium like Google Voice) allow them to escape the confines of the Network and bridge in outsiders seamlessly and give them as much of the experience as their devices would allow - think a WebRTC Video Conference with some people listening in voice only all the way down to SMS users getting a stream of Voice-2-Text transcription of the conversation and the ability to respond, that being converted via Text-2-Speech back to the listening participants.

So yeah, I have been thinking about possibilities!

Many people already donde that, but since WebRTC is not an RFC yet, google change many things and only Chrome and firefox works your option right now is use a mediagateway.


Webrtc can be done now
Sip video can be done now
Sip audio cab be done now
Pstn to cell can be done now
SMS can be done now
Email typically doesn’t use audio or video codecs and neither does SMS so no idea why they are here…

The question was maybe too vague…
What does VP8 give you that you Dont already have. The answer is other than a new codec, nothing