Klaus Darilion wrote: > Thus, if I have 3G video call bridged by Asterisk from chan_zap to > chan_misdn Asterisk should decode the 3G video and then reencode it - > does not sound well engineered. > > If Asterisk bridges a G4 fax call from chan_zap to chan_misdn Asterisk > should decode the G4 fax into in an IMAGE and the reencode it to G4?
If you don't, then you are really defeating the purpose of using Asterisk in the first place. Let me list a few small examples that would work if you do break the stream down into voice and video frames, but will not work if you pass the stream through as raw. There are countless more ... 1) What if you want to monitor the call via ChanSpy? Specifically, what if you want to monitor the call from a channel type that is not an ISDN channel? 2) What happens if one caller wants to transfer the other to a non-ISDN channel? 3) What if you want to transfer the channels into a conference with other ISDN channels? (Manager action redirect, for example) I understand your desire to make this work. I can also appreciate you seeing the "easy way" to do it, and pushing that direction. However, what I and Tilghman have been trying to do is push this more toward the "Asterisk way" of doing this. Asterisk is designed to be a channel technology independent application. That means, it does not matter if it is ISDN or SIP or whatever else, it all looks the same inside of Asterisk. This lets us build features that will work across any type of call. What you are proposing completely breaks this architecture. So, for a little bit of extra work and minor effect on performance, you can make this fit into the Asterisk architecture, bridging 3G ISDN voice and video calls to everything else that Asterisk has to offer. -- Russell Bryant Software Engineer Digium, Inc. _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
