[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
with
> 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
a
> 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
itself.

Richard




More information about the Standards mailing list