Have you been confirmed whether SIP UDP Ports are not blocked by ISP? This
is very common.
Try to use some other UDP port which probably should not be blocked by ISP.
Best Regards,
Vivek Batra
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementor
Hi All,
I have a (basic?) query regarding inter-domain SIP calls.
SIP softphone1 (Xlite) is running on a machine connected to the internet via
ISP1 (Internet Service Provider 1). The proxy of this softphone 1 is running on
a machine which has a fixed IP a.b.c.d (provided by ISP1).
SIP softp
This timer is to allow for the possibility of resent request to match up with
the transaction that just "completed". If we didn't do this, a subsequent
final response to the re-send could be different than that of the initial
final response.
So why do you need this for an "unreliable" transport?
Hi all,
In Section 17.2.2, following is mentioned:
When the server transaction enters the "Completed" state, it MUST set
Timer J to fire in 64*T1 seconds for unreliable transports, and zero
seconds for reliable transports. While in the "Completed" state, the
server transaction MUST pa
On Fri, Oct 23, 2009 at 3:07 PM, Manoj Priyankara [TG]
wrote:
> Dear All,
>
> How we can find out the best value of the jitter buffer for a VoIP end
> point? Any recommendations for the same?
Take a look at this:
http://books.google.es/books?id=hT60-P-CSF4C&pg=PA102&lpg=PA102&dq=buffer+length+L+j
Hi Anna
ACK should be ignored by UAS as ACK is only sent in response to response of
INVITE request.
Same can be referred from section 4 of RFC 3261 which mentions "Alice confirms
receipt of the BYE with a 200
(OK) response, which terminates the session and the BYE transaction. No ACK is
sent -
42
2009/10/23 Manoj Priyankara [TG]
> Dear All,
>
> How we can find out the best value of the jitter buffer for a VoIP end
> point? Any recommendations for the same?
>
> Thanks
> BR,
> Manoj
>
> ___
> Sip-implementors mailing list
> Sip-implementors@li
El Martes, 27 de Octubre de 2009, Gellatly, Anna escribió:
> Hello Sip List -
>
> What should a UAS do when an ACK is received out of dialog - ignore or
> send 481 Transaction does not exist?
Ignore it, as ACK has never response and the UAC would never expect a response
for an ACK.
--
Iñaki Ba
Hi Anna,
The ACK should be ignored. An ACK is only sent in response to a response to an
INVITE request.
Thanks,
Alok Tiwari
Aricent
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Gellatly,
Ann
Hello Sip List -
What should a UAS do when an ACK is received out of dialog - ignore or
send 481 Transaction does not exist?
Thanks,
Anna.
___
Sip-implementors mailing list
Sip-implementors@lists.cs.columbia.edu
https://lists.cs.columbia.edu/cucsli
Howdy,
We have a SIP Proxy implementation and wanted to check what are the
specific requirements for a (GSMA) IPX SIP Proxy. I've found PRD IR.34
(Version 4.8, 29 September 2009) but I'm not sure this is the right
reference. While Annex B lists a number of requirements, it's not
really clear to me
El Martes, 27 de Octubre de 2009, Laurent Etiemble escribió:
> Hello,
>
> Cross-domain access goes far beyond the RFC. Such situations are well
> covered by 3GPP and OMA (they call it roaming). In the case of XDM,
> the only releated specification I know is the GSMA IR.90 (chapter 4):
> they are d
Hello,
Cross-domain access goes far beyond the RFC. Such situations are well
covered by 3GPP and OMA (they call it roaming). In the case of XDM,
the only releated specification I know is the GSMA IR.90 (chapter 4):
they are describing an element named a cross-network proxy that acts
as a single po
13 matches
Mail list logo
