[standards-jig] VoIP Standard for Jabber

Richard Dobson richard at dobson-i.net
Wed Feb 5 16:50:55 UTC 2003

> I have been doing some research into many of the possibilities for VoIP
> Jabber and it seems to me that XMPP really would be the best option for
> session initiation.  Here are my reasons.  First, XMPP is easily Flexible
> enough to handle all of the same capabilities that SIP offers in this
> department, and may be able to do it without the same level of chat.
> Second,  XMPP is already Jabbers engine and can probably be easily
> integrated into Jabber.  Third, it would be easier for clients to adopt an
> XMPP based solution.
> If we push for SIP in an XMPP environment, then we need to develop a
> protocol for handing XMPP off to SIP for the porpose of Voice, and client
> will then have to support both SIP and XMPP, both of which are session
> protocols.  And if it really is easy to bridge the SIP and XMPP protocols,
> then do it via a Gateway or Transport so non XMPP based clients can handle
> it as well.

Yep seems fine to me

> RTP should be used for actually carrying the voice and I think it would be
> wise for the Jabber community to adopt a number of standarized codecs and
> few not quite so standard codecs.

Speex seems a good codec to use, open source and patent-free.

> Also, one thing I think would be benificial would be the ability to change
> compression/codec in the middle of a call.  This would provide for a level
> of tolerance to poor bandwidth environments.  I know this is possible as
> Teamspeak supports then and it works quite well, in the middle of a
> conference, the rooms codec can be changed if needed without causing the
> client to disconnect and reconnect.

Certainly a possibility, does anyone know if RTP can handle this itself?
Also speex can do variable bit rate which could solve this problem all by


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