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.
ID TYPE NAME DESCRIPTION
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.
Like:
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