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

[Mike Albon]

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