[JDEV] re: Regarding adding Voice support in Jabber Client.

[Stephen D. Williams]

For voice, I may agree somewhat, however there are times when even this 
is useful.  When you are blocked by firewalls, NAT, etc., it can be 
acceptable, especially when 8kb is tiny compared to available bandwidth.

A better use for bulk data through the Jabber server, for certain 
situations, would be for:

   * Video/Image snapshots  (Instant Images that I implemented at AOL)
   * Bulk data transfer (peer to peer, etc.)
   * Application to application XML traffic.

[Hi Jabber community, I'm back!  ;-)  Did I miss much??  ]

sdw

Mike Albon wrote:

>Hi All,
>
>I am not currently on the list but read the archives. Can I say I am
>quite concerned about the thought of encoding voice and sending as a
>payload over the XML stream. 
> 
>Latency and TCP are going to cause problems with Voice traffic. TCP is a
>good protocol if you don't want data to go missing, but it is not ideal
>for real-time services like voice/video conferencing. As you have to go
>via at least one Jabber server with 8 Kbits (or more depending on Codec)
>of voice turns into a larger payload if you have to encode it and
>packetise it. Ideally for voice using between 32 and 64 samples per
>packet provides reasonable voice and smaller packets reduce echo (which
>is why ATM uses 48 byte payloads). This will put loads of extra load on
>the Jabber Server and links. These services usually use something like
>h323 or RTP to transport the voice information.
>
>Suggestion:
>
>Use the jabber:iq:oob namespace (or similar) to provide a SIP url, this
>would enable Voice over IP (VOIP) clients to communicate properly. It
>also provides your jabber client with an opportunity to connect with MSN
>clients of the future as Microsoft has committed to go with SIP. (The
>next version of NetMeeting will support it). Also a lot of the phone
>companies are looking to SIP to solve the limitations with H323. In
>jabber a client like an MSN chatter could then connect to the servers
>SIP gateway and redirect the call to the correct recipient if required
>or the real address transmitted directly. So to connect to me it would
>be something like sip://mikea@hopeless-newbie.co.uk.
>
>SIP is available under Unix with the Linphone and Vovida VOCAL projects.
>VOCAL has a console client which would probably be very easy to
>integrate.
>
>I certainly don't wish to preempt the people working on the JEP with my
>comments but please don't pass voice through the server.
>
>Mike Albon
>
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