[asterisk-users] SIP/TCP?

[Yuan LIU]

>From: "James R. Stevens" <jstevens at athensdistributing.com>
>
>TCP is a connection oriented protocol ..as others mentioned, it superiority 
>comes because it knows when packets are dropped to resend them. It also has 
>mechanisms for flow control etc.. SIP is a connection-less protocol. It 
>uses 'best effort' transmissions..if u want its delivery guaranteed you 
>must encapsulate it.

So I take it that UDP is just a decision due to popular demand; timing 
(jitter) is a frequently cited factor to favour UDP.  Is there any technical 
difficulties in implementing SIP/TCP within Asterisk?

The reason I'm asking is that there are products that support both UDP and 
TCP.  And SIP/TCP, RTP/TCP have their own merits.

Granted, SIP is connectionless.  So is HTTP (well, for its original design 
anyway).  I notice that guaranteed delivery could be a good thing for SIP in 
many situations; there have also been advancements that make timing less an 
issue in RTP/TCP.

Is "switching to" SIP/TCP - RTP/TCP as simple as rewrap messages/streams, or 
is it more involved?

Yuan Liu

>-----Original Message-----
>From: "Yuan LIU"<yliu11 at hotmail.com>
>Sent: 1/5/07 1:26:38 PM
>To: "asterisk-users at lists.digium.com"<asterisk-users at lists.digium.com>
>Subject: [asterisk-users] SIP/TCP?
>
>I'm still learning some of the basics.  Can someone explain in layman's
>terms what's the difficulty for Asterisk to support SIP/TCP (and even
>RTP/TCP)?