[asterisk-dev] Using RTP in my own channel driver

[Henning Holtschneider]

Hello everybody,

I'm using Asterisk on an embedded device which features two FXO ports.  
I'd like to use those ports from Asterisk and I'm currently trying to  
figure out the best approach to do so.

Unfortunately, the API provided by the device manufacturer only allows  
me to read and write RTP packets from and to device nodes representing  
the FXO ports. I've come up with four possible solutions:

1. Strip off off RTP headers coming from the device before sending the  
payload into Asterisk via ast_queue_frame(). When writing data to the  
device, prepend (fake) RTP headers to the data from the ast_frame.

2. Write a codec translator which translates between the raw audio  
payload and the payload with RTP headers. This is essentially the same  
as 1., except that the translation is taking place outside of the  
channel driver itself.

3. If the other leg of the channel uses RTP, create a direct RTP  
bridge. I doubt this is currently possible in Asterisk but I've found  
references to work-in-progress on external RTP handling (<http://lists.digium.com/pipermail/asterisk-dev/2008-September/034367.html 
 >). Can anyone give me some information on the current status of this  
functionality?

And last but not least:

4. Write an external application that does the FXO port handling and  
talk to Asterisk via SIP. The FXO ports would simply act as SIP peers  
to Asterisk.

The last solution does not have anything to do with Asterisk directly,  
but I wonder if this is the most effective approach if there is no  
good way to use RTP from inside Asterisk. I'd appreciate your feedback  
on this. I would also welcome any advice if there are other ways to  
achive my goal that I haven't thought about yet.

Best regards,
Henning Holtschneider
--
LocaNet oHG - http://www.loca.net
Lindemannstrasse 81, D-44137 Dortmund
tel +49 231 91596-25, fax +49 231 91596-55
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Geschäftsführer Sven Haufe, Henning Holtschneider

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