[asterisk-users] SIP over TCP

[Asterisk]

Hi,

But you can only route SIP signalization over TCP. Audio stream must still go thru UDP, right?

BR, Alex

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kristian Kielhofner
Sent: Sunday, June 22, 2008 4:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP over TCP

On 6/22/08, Michael Graves <mgraves at mstvp.com> wrote:
> Ok, so now that it's possible to implement SIP over TCP instead of UDP
>  why would I want to do this? Beyond simply integration with M$ OCS.
>
>  And what are the implications for management of QoS? I would expect
>  that lost packets would be less of a factor.
>
>  Thanks,
>
>  Michael
>  --
>  Michael Graves
>  mgraves<at>mstvp.com
>  http://blog.mgraves.org
>  o713-861-4005
>  c713-201-1262
>  sip:mjgraves at pixelpower.onsip.com
>  skype mjgraves
>  54245 at fwd.pulver.com
>

Michael,

  The main advantages for SIP over TCP that I know of (in no particular order):

- Better compatibility with NAT devices (it seems some of them don't
do UDP well)
- Support for TLS
- Support for packet fragmentation (to support large/diverse SDPs, headers, etc)

  I'm sure there are other ones but that's all I can think of this
early on a Sunday morning...


--
Kristian Kielhofner
NOT sent from my iPhone or Blackberry

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