At 04:34 AM 4/25/2008, Donald Lee wrote:in case there is firewall or like nat issues b/w C and B, the normal REFER will be failed to enable the C INVITE B. how the INVITE can be routed to B correctly?On Fri, Apr 25, 2008 at 2:18 PM, ajay thakur <<mailto:thakur.ajay.ietf at gmail.com>thakur.ajay.ietf at gmail.com> wrote:
James If there is any kind of firewall enabled on Router then connection
from C to B is not allowed. That's what is main worry :(
First of all, no one mentioned a firewall in this scenario.
But if there is a firewall, how exactly will the firewall fail the SIP request (I know it can be configured to block anything (or everything), but is this normally the problem)? Won't SIP perform fine, and the problem be a RTP stream issue *IF* there is a NAT involved?
A lot depends on where the firewall (and NAT) are in this A-B-C scenario below.
BTW - check this ID out for answers to NATs (that weren't offered in the original scenario):
Title : Best Current Practices for NAT Traversal for SIP
Author(s) : C. Boulton, et al.
Filename : draft-ietf-sipping-nat-scenarios-08.txt
Pages : 62
Date : 2008-04-25
Traversal of the Session Initiation Protocol (SIP) and the sessions
it establishes through Network Address Translators (NAT) is a complex
problem. Currently there are many deployment scenarios and traversal
mechanisms for media traffic. This document aims to provide concrete
recommendations and a unified method for NAT traversal as well as
documenting corresponding flows.
A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-nat-scenarios-08.txt
AjaySip mailing list <https://www.ietf.org/mailman/listinfo/sip>https://www.ietf.org/mailman/listinfo/sip
On Fri, Apr 25, 2008 at 5:33 AM, James M. Polk <<mailto:jmpolk at cisco.com>jmpolk at cisco.com> wrote:
At 04:04 AM 4/24/2008, ajay thakur wrote:
Hi guys,
I have one doubt related to refer method.
A---------|
|
|--------Router-----------------C
B---------|
In the above topology A & B phones are internal phones. C is external phone.
Scenario is A calls C and then A refers B to C;
If A calls C, the either A or C need to send the REFER to the other, telling one of them to contact B. For example, A calls C, and then A sends REFER to C to contact B. C will now send an INVITE to B with a Contact: header value of A
So when A sends refer message to C it includes B's SIP URI.
In the Refer-To: header, yes
Now my doubt is
can we have port number along with this address.
why not?
e.g. Refer-To:<sip:xxx at ip-address:port-number>
Thanks
Ajay
_______________________________________________Use <mailto:sip-implementors at cs.columbia.edu>sip-implementors at cs.columbia.edu for questions on current sip
This list is for NEW development of the core SIP Protocol
Use <mailto:sipping at ietf.org>sipping at ietf.org for new developments on the application of sip
_______________________________________________
Sip mailing list <https://www.ietf.org/mailman/listinfo/sip>https://www.ietf.org/mailman/listinfo/sipUse <mailto:sip-implementors at cs.columbia.edu>sip-implementors at cs.columbia.edu for questions on current sip
This list is for NEW development of the core SIP Protocol
Use <mailto:sipping at ietf.org>sipping at ietf.org for new developments on the application of sip
--
BR
Donald
_______________________________________________ Sip mailing list https://www.ietf.org/mailman/listinfo/sip This list is for NEW development of the core SIP Protocol Use sip-implementors at cs.columbia.edu for questions on current sip Use sipping at ietf.org for new developments on the application of sip