Posted: Fri Jun 09, 2006 12:38 am Post subject: [Asterisk-video] H320
Olle E Johansson wrote:
Quote:
> AFAIK, H324M and H320 just open a b channel of type Unrestricted
> digital information (instead of the Speech that it's the one that
> asterisk does), and on top of that channels it starts negotiating
> the protocol. So having the isdn channel abstracted is a good thing
> as you don't need to handle the all the isdn internalls.
> All we need is an isdn channel that offer a clean digital channel and
> does the correct q931 negotiation.
Which is done in chan_zap - or?
It's done in libpri but with parameters set by chan_zap, I have just hardcode the q931 setup message and I can make outgoing videocall now.. :)
Quote:
>
> The idea of the application (for incomming calls for example) I have
> is one like this:
>
> exten => s,1,Answer
> exten => s,2,H324MTransfer(SIP/blablabla)
Why not use dial?
I don't know if Dial can be used in some way to do what I need.
I just need to channels an isdn raw digital one and another with audio/video support, if dial can just make an outgoing call and pass both channels to another application it would work perfectly.
Quote:
>
> So you've got an isdn channel in the application with h324m, in which
> you start the h324m negotiation.
Which may belong in the channel driver...
No, the h324m negotiation is done on top of an established b-channel.
Just like a fax is negotiated on top of a pstn call.
Quote:
> You read from the channel pass it to the h324m code and viceversa.
> Once you have the call stablished, dial the sip peer and start
> reading the audio and video from the channle and passing it as
> input to the h324 which will mux it and send it to the isdn part,
> and continue reading from the isdn channel demuxing it and passing
> the audio/video to the sip channel.
But we can't establish a call until we know we have something in the
other end that can accept it.
We might have to send the call to voicemail or forward it somewhere
else before we answer or simply not answer at all, because
no one is available... We don't want to re-implement app_dial, it's
already been done once... ;-)
I agree.
What h324m-SIP gateways typically do is a kind of third party call negotiation.
When it gets an incoming call it sends an empty INVITE to the sip peer.
If it receives a 200 with the sdp it continues the h324m negotiation with that capabilities (or with the ones he can transcode).
Later, when the call is successfully established with the mobile phone then it sends the ACK with the SDP to the sip client.
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Posted: Fri Jun 09, 2006 1:26 am Post subject: [Asterisk-video] H320
Wasn't there a patch somewhere to allow data calls between mISDN and zaptel?
Maybe that patch could help somehow.
Christian K.
(not to be confused with Christian Richter)
Quote:
-----Original Message-----
From: asterisk-video-bounces@lists.digium.com
[mailto:asterisk-video-bounces@lists.digium.com] On Behalf Of
Olle E Johansson
Sent: Freitag, 9. Juni 2006 11:57
To: Development discussion of video media support in Asterisk
Cc: Christian Richter
Subject: Re: [Asterisk-video] H320
Thank you all for the explanations!
Seems like it is a channel that needs it's own interface to
the zap drivers, then.
We will have to talk to Christian about mISDN and see how
clever we can be.
(Christian - check the mail archives for asterisk-video to
know what kind of discussion I just involved you in :-)
/Olle
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Posted: Fri Jun 09, 2006 1:42 am Post subject: [Asterisk-video] H320
Quote:
I think H324M and H320 would have simillar issues. First we would need to change libpri/zaptel in order to allow changing the q931 to support more types of call than just speech ones.
I also found that it could be interesting to change the default frame size for reading from b channels.
I have added a Patch #6251 to allow using HDLC Decoding for bchannels
for chan_zap, that might be useful in that case, I think the video data
is transported in hdlc
?
Anyway i have changed the bchannel readsize in that patch too..
Posted: Fri Jun 09, 2006 1:43 am Post subject: [Asterisk-video] H320
Hi H320 guys :)
I have some customers that do video passthrough with chan_misdn, which
works quite OK for them. So as mentioned before H320 is in the
bchannels, and is therefore neither very interesting for chan_zap nor
for chan_misdn.
Things that happen in the voice path are only intersting for apps.
What you want to do happens already with audio data, lets say you have a
sip phone which uses codec g729 and call over chan_misdn to the pstn
which knows only alaw/ulaw.
In the audio case we call the thing codec that does the magic, why
shouldn't we have codecs that transcode between h.320 and the sip/h323
standards ?
The only problem here is that the codecs work only on single voicepaths,
we would need to tell the codecs that there might be other voicepaths
which are used for bonding, but the channeldrivers like chan_misdn and
chan_zap know which number and which type of calls are involved and
could tell the codecs which voice-paths might be used for bonding.
Posted: Fri Jun 09, 2006 4:04 am Post subject: [Asterisk-video] H320
That's the point, we work with video conference from Aethra, a italian
company, the point is to turn asterisk into a gateway IP to ISDN, being
able to bond isdn channels to a better video quality, and this equipment
from radvision or tandberg are expensive, and a lot of company use ISDN.
Sergio Garc?a Murillo wrote:
Quote:
Olle E Johansson wrote:
> 8 jun 2006 kl. 22.40 skrev Ian Esper:
>
>
>> I was thinking the possibility to incoporate H320 in asterisk, what
>> are the possibilities on doing that? Was it discussed before?
>>
>>
> Can you please explain a bit more? Why H320 - what's the application?
>
If I recall correctly, H320 was the protocol for ISDN videoconferencing.
It's a bit tricky to implement since you can use more than one B channel for the same call (bonding).
The H320 videoconference kits are (or used to be) very expensive, but the quality of the video was great also. For that I think that many companies are still using it and don't plan to change it in the near future.
So it could be great to have it integrated into asterisk, so we could offer a full IP(SIP,H323)-ISDN(H320)-UMTS(H324M) gateway. And make everything work with meetme conferences :)
Greetings
Sergio
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