[Asterisk-Users] Spring VON Wrap Up

[Jan Janak]

On 05-04 14:35, Steven Sokol wrote:
> 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.

  There are some more reasons -- transport of big SIP messages and
  avoiding network congestion among them. SIP message can get pretty big
  when XML encoded documents (presence documents, for example) are
  attached.

  TCP does not fit everywhere. It is still advantageous to let SIP
  phones use UDP when communicating with a proxy because the proxy does
  not have to keep a list of opened connections which is very resource
  consuming (just consider that you have 100000 users using the same
  proxy -- that can be easily achieved using single server).

  On the other hand, TCP is useful for proxy-to-proxy communication,
  especially when there is bigger amount of traffic between proxies. In
  this case TCP head blocking is really not a problem because the sender
  gets constant feedback from the remote party and can retransmit the
  lost segment in a short time. (There was a technical report on this 
  published by Henning Schulzrinne).

    Jan.