Hi, But you can only route SIP signalization over TCP. Audio stream must still go thru UDP, right?
BR, Alex -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] > skype mjgraves > [EMAIL PROTECTED] > 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 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
