[Asterisk-video] H320

[Sergio García Murillo]

Olle E Johansson wrote:
>> 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.. :)

>> 
>> 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.

>> 
>> 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.


> 
>> 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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