[Asterisk-Users] Spring VON Wrap Up

[Steven Sokol]

> On Apr 5, 2004, at 12:10 PM, Olle E. Johansson wrote:
> > Members of the IETF added information on the to-be-standardized
> > standard,
> > meaning that SIP with TLS over TCP will be mandatory. We need to start
> > working
> > on TCP and TLS support.
> 
> Could someone explain to me why anyone in their right mind would ever
> want to run VoIP (or any lossy real-time data) over TCP?  Unless I'm
> missing something, the effects of packet loss would be almost perfectly
> pessimal.  Every time you lose a packet, the receiver stalls and then
> can't catch up, so you get horrifically huge delays.  Does it actually
> gain something for anyone doing voice or video?
> 

TCP/TLS would be used for the SIP messaging which handles call setup,
teardown, and other non-Realtime functions.  The voice stream will still be
handled via RTP which is a UDP-based protocol.

The reason for doing the call setup as TCP is to allow for TLS encryption.
The SIP messages themselves are simply bits of ASCII text (much like SMTP
messages).  Currently Asterisk does SIP over UDP only (I think...).  In
order to support SIPS (Secure SIP, like HTTPS) we need to build a version of
chan_sip (or chan_sip2 ;-) that supports SIP over TCP.  The voice stream
will remain UDP an therefore not succumb to enormous delay.

-S