[on-asterisk] skype, iax, sip and silk

[Terry D. Cudney]

Hi all,

Reading about skype - asterisk and skype - freeswitch leaves me wondering about the various levels of interaction/compatibility. Maybe someone here can explain it for us.

This is how I understand it. (Please turn your flame-throwers down to "warm" and use them sparingly.) Please corect me and or expand on it as appropriate:

SILK is the skype codec for higher bandwidth, higher fidelity communications that has recently been made publicly available.

Can SILK be carried over SIP(or IAX)/RTP connections? like g711/g729/etc? or is there some other protocol requirement that ties SIL to the skype code?

If SILK can be used as just-another-codec with SIP(IAX)/RTP, then it could provide higher fidelity than the other codecs currently being used with asterisk/freeswitch, right?

Connectivity with skype end-points is a different issue which will require integration of skype protocols (at least via their API) with asterisk/freeswitch.

From links followed from Wes's posting yesterday, it looks like work is underway to bring about this integration.

The TAUG beta of Skype for Asterisk is an example of this coming integration, right.

I do not have skype, but it looks like it'd be worth checking out. Simon, how has the TAUG beta been doing? Is it still available for testing?

Thanks to all who can add to /correct these ramblings,

--terry

--
Name: Terry D. Cudney
Phone: (705) 812-4949
SIP: 50...@cudney.homeip.net
E-mail: te...@octothorp.org

Please avoid sending me Word or PowerPoint attachments.
See http://www.gnu.org/philosophy/no-word-attachments.html

Having a smoking section in a restaurant is like...
having a peeing sectionin a swimming pool.

---------------------------------------------------------------------
To unsubscribe, e-mail: asterisk-u...@uc.org
For additional commands, e-mail: asteri...@uc.org

[Simon P. Ditner]

It's still available for testing, but it's not getting much use.

To my knowledge, you're correct, silk is simply a codec that can be
carried over any old RTP stream, and it's a matter of endpoints
supporting it, or transcoding to G722 for it to be useful.

Cheers,
spd