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mgraves at mstvp.com
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 Posted: Sun Jun 22, 2008 9:21 am Post subject: [asterisk-users] SIP over TCP |
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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 |
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kkielhofner at star2st...
Guest
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| Posted: Sun Jun 22, 2008 9:49 am Post subject: [asterisk-users] SIP over TCP |
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On 6/22/08, Michael Graves <mgraves at mstvp.com> wrote:
| Quote: | 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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asterisk at abraxas.si
Guest
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| Posted: Mon Jun 23, 2008 9:59 am Post subject: [asterisk-users] SIP over TCP |
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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:
| Quote: | 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 |
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mgraves at mstvp.com
Guest
|
| Posted: Mon Jun 23, 2008 6:28 pm Post subject: [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:
| Quote: | 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:
| Quote: | 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 |
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asterisk at abraxas.si
Guest
|
| Posted: Tue Jun 24, 2008 5:36 am Post subject: [asterisk-users] SIP over TCP |
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That's excellent! So in theory one could not make Asterisk compatible SIP softphone in Flash (since Flash only supports TCP). Nice...
BR, Alex
-----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:
| Quote: | 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:
| Quote: | 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 |
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mgraves at mstvp.com
Guest
|
| Posted: Tue Jun 24, 2008 7:52 am Post subject: [asterisk-users] SIP over TCP |
|
|
On Tue, 24 Jun 2008 12:36:52 +0200, Asterisk wrote:
| Quote: | 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.
| Quote: |
-----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:
| Quote: | 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:
| Quote: | 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 |
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benny+usenet at amorse...
Guest
|
| Posted: Tue Jun 24, 2008 5:48 pm Post subject: [asterisk-users] SIP over TCP |
|
|
"Michael Graves" <mgraves at mstvp.com> writes:
| Quote: | No, TCP for media as well. I though that was the whole point of SIP
over TCP.
|
Hopefully not. RTP over TCP would be entirely pointless. RTP needs
packetization, doesn't mind packet loss (within reason) but hates
retransmissions. TCP doesn't provide packetization, guarantees against
packet loss, but retransmits.
/Benny |
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