[asterisk-users] SIP over TCP

[Michael Graves]

On Tue, 24 Jun 2008 12:36:52 +0200, Asterisk wrote:

>That's excellent! So in theory one could not make Asterisk compatible SIP softphone in Flash (since Flash only supports TCP). Nice...
>
>BR, Alex

I beleive that this hs already been done, although I can't recall by
whom.

>
>-----Original Message-----
>From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Michael Graves
>Sent: Tuesday, June 24, 2008 1:29 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [asterisk-users] SIP over TCP
>
>No, TCP for media as well. I though that was the whole point of SIP
>over TCP.
>
>Michael
>
>
>On Mon, 23 Jun 2008 16:59:00 +0200, Asterisk wrote:
>
>>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
>>
>>_______________________________________________
>>-- 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
>>
>
>--
>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
>
>
>
>_______________________________________________
>-- 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
>

--
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