Please see:
http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-00.txt
which discusses NAT traversal for SIP signaling. Bindings for UDP are
kept alive by sending STUN packets to the SIP port of the SIP server.
The document talks about CRLF for TCP, but per IETF 63 this will also
change to STUN over TCP.
Keeping NAT bindings alive for media traffic is covered in ICE:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-ice-05.txt
The approach there uses STUN sent to the RTP ports when the peer
supports ICE, otherwise recommends RTP no-op:
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-no-op-00.txt
-Jonathan R.
Paulo de Arruda Borelli wrote:
Hi, Jun,
Are you aware of any recommendation for SIP over UDP?
-----Original Message-----
From: Hyoungjoon Park [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 16, 2005 7:11 AM
To: [EMAIL PROTECTED]
Cc: [email protected]
Subject: RE: [Sip-implementors] keep alive with NAT in SIP
Hi, Paulo,
I think IETF has recommended the CR/LF mechanism for SIP over TCP, although
I'm quite not sure for NAT case.
Please check out the following link;
http://www.softarmor.com/sipwg/meets/ietf61/notes/minutes-sip-ietf61.html
"1. TCP keepalive every periodic number of seconds? CRLF?
REGISTER? New message (PING)?
Comments:
- REGISTER is horrifically expensive - not a general solution
- CRLF doesn't generate an application ACK - have to wait for TCP
keepalives. question about if standard TCP interface allows check. We
think we can check unacked bytes in most(?) socket interfaces and use CRLF.
"
Regards,
Jun
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paulo de
Arruda Borelli
Sent: Saturday, August 13, 2005 4:00 AM
To: [email protected]
Subject: RE: [Sip-implementors] keep alive with NAT in SIP
Hi, Rakesh,
You can use the expires value of the REGISTER messages. Just configure the
proxy to require a low expires value (like 60 seconds). This will force the
endpoint o re-register again every 60 seconds. In fact, in less than 60
seconds - because, if they re-register exactly every 60 seconds, they will
certainly lose registration until the new registration gets active. Some
endpoints will re-register at 50% of expires (30 seconds, in this case);
some will at 90%. But they usually re-register at less than the nominal
expires value.
The idea is that this will force the NAT pinhole to be open all the time -
enabling inbound flow (from the proxy to the endpoint) to reach the
endpoint.
Some brands (of endpoints) also support the OPTIONS method as keep-alive
mechanism. This will also keep the NAT pinhole open. But the re-register
method is great - since it's usually supported. Notice it's imperative that
the proxy force expires to a low value. Usually, the endpoints send a
default value of 1800 or 3600 seconds - and not always configurable. To keep
the NAT pinhole open, it's imperative to re-register at less than one
minute.
Paulo Borelli
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nataraju A B
Sent: Friday, August 12, 2005 5:37 AM
To: 'Rakesh Dhandhukiya'; [email protected]
Subject: RE: [Sip-implementors] keep alive with NAT in SIP
Regards,
Nataraju A.B.
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rakesh
Dhandhukiya
Sent: Tuesday, August 09, 2005 1:03 PM
To: [email protected]
Subject: [Sip-implementors] keep alive with NAT in SIP
Hi !
I want to keep alive with NAT. First of how to know the keep alive timeout
of NAT for UDP. Is there any file like tcp timeout ?
Is ICE used for NAT keep alive?
[ABN] service provider may not disclose these timer values... only way to
find out these timer values would be to use trail and error method....
I don't think ICE been used to NAT keep alive...
Thanks.
Rakesh Dhandhukiya
_______________________________________________
Sip-implementors mailing list
[email protected]
http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
_______________________________________________
Sip-implementors mailing list
[email protected]
http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
_______________________________________________
Sip-implementors mailing list
[email protected]
http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
_______________________________________________
Sip-implementors mailing list
[email protected]
http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
--
Jonathan D. Rosenberg, Ph.D. 600 Lanidex Plaza
Director, Service Provider VoIP Architecture Parsippany, NJ 07054-2711
Cisco Systems
[EMAIL PROTECTED] FAX: (973) 952-5050
http://www.jdrosen.net PHONE: (973) 952-5000
http://www.cisco.com
_______________________________________________
Sip-implementors mailing list
[email protected]
http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
