Hello Folks.
SIP faces with overload issue recently due to its popularity. Build-in
overload mechanism cannot prevent overload effectively. Therefore, IETF SIP
Overload Control (soc) Working Group works on IETF RFC "SIP Overload Control"
currently.
http://tools.ietf.org/html/draft-ietf-soc-o
On 7/16/13 3:33 PM, Joel Gerber wrote:
> Sorry; dialog being modified wasn't the most accurate terminology. The way I
> understand the workings in the Genband environment, the subsequent (and
> preceeding) digits are sent in an INFO message either during the provisional
> stage of dialog setup o
Sorry; dialog being modified wasn't the most accurate terminology. The way I
understand the workings in the Genband environment, the subsequent (and
preceeding) digits are sent in an INFO message either during the provisional
stage of dialog setup or after the dialog is fully established. I thin
The race condition is known. I assume that the answers can be found at
https://lists.cs.columbia.edu/pipermail/sip-implementors/2010-October/025881.html
which indicates to look at
https://lists.cs.columbia.edu/pipermail/sip-implementors/2010-May/025009.html
http://www.ietf.org/mail-archive/web
On 7/16/13 1:41 PM, Joel Gerber wrote:
> I believe it actually establishes the dialog with the partial digits. Then
> the dialog is modified with the new digits as they are received. I haven't
> tested this in a lab, so I can't be 100% sure, but that is what I've been led
> to believe.
What do
Hi,
I have a call flow as shown below.
UAC--UAS
INVITE with SDP --->
< 183 with SDP
< UPDATE without SDP
< 200 INVITE with SDP
200 UPDATE without SDP ->
Here, there is negotiation of Sess
I believe it actually establishes the dialog with the partial digits. Then the
dialog is modified with the new digits as they are received. I haven't tested
this in a lab, so I can't be 100% sure, but that is what I've been led to
believe.
As to your question about how many digits must be sent
The Genband C20/CS2000 switch has an element called an SST that does the PSTN
to SIP interworking. This is the device that supports using INFO messages for
overlap dialing. It is acting as a gateway, hence its ability to send out the
INFO messages during, I believe (I’m not 100% sure that it’s n
The problem with the INFO method is that you first must establish a
dialog with *something*, and you need a URI do do that. And once you
have established that dialog, all the digits you send with INFO are
going to it.
So this really only works with certain topologies, and with the calling
devi
Is Genband's C20/CS2000 switches used as a gateway to PSTN and SIP network? or
is it used as a pure SIP network element (proxy, B2BUA, etc)?
If *only SIP is involved*, I can *not* figure out how the INFO method can be
used to convey overlapping digits. INFO is sent within an early dialog create
Hi,
RFC 3966 section 3 indicates the following:
"Characters other than those in the "reserved" and "unsafe" sets (see
RFC 2396 [RFC2396]) are equivalent to their "% HEX HEX" percent
encoding."
Is the statement true only where the ABNF allows "pct-encoded"? For instance,
is it valid to useles
I know Genband's C20/CS2000 switches support (and default) to using the INFO
method for overlapping digits (this is what they call partial dialled digits
followed by subsequent digits).
Joel Gerber
Network Specialist
Network Operations
Eastlink
E: joel.ger...@corp.eastlink.ca T: 519.786.1241
--
Thanks to all!
I found one internet draft that propose to use the INFO method to convey
subsequent dialed numbers:
http://tools.ietf.org/id/draft-zhang-sipping-overlap-01.txt
It claimed to resolve the issues related to the INVITE/484/ACK approach in
RFC3578, but this draft seems to be decea
> Need inputs in the following scenario (wrt UAS)
>
> - INVITE received with Session-Expires, Min-SE, Supported:timer, no
> refresher is present
> - 183 sent with Supported:timer
> - UPDATE received with *no* Session-Expires, *no* Min-SE,
> Supported:timer, no refresher is present
> - 200 OK of UP
> In my opinion, if only a SIP network is involved and
> no gateways are used, overlap signalling (e.g., the
> caller sends dialed digits to an outbound proxy in
> consecutive separate INVITEs for the outbound proxy
> to collect enough information and route the requests)
> is meaningless, becau
A lot of the time you wouldn't be able to find out that a wrong number was
dialled until all of the digits were collected.
Joel Gerber
Network Specialist
Network Operations
Eastlink
E: joel.ger...@corp.eastlink.ca T: 519.786.1241
-Original Message-
From: SIP Learner [mailto:rfc3...@foxmai
I will add one point to this discussion; sometimes not sending out all the
digits at once makes sense, but not with SIP.
Gateway protocols like MGCP will commonly send out digits one at a time. This
allows a Gateway Controlled protocol to allow the controller to decide when
enough digits have b
> I think there is one benefit
> *when the caller dialed a wrong number!*
>
> If overlap signallig were used, the caller might be
> informed by the proxy immediately. In en bloc
> signalling, however, the caller will only be altered
> after he dialed the complete wrong number
> watsting time
Hi,
My reading is that 200 OK to INVITE with Require:timer but no
Session-Expires is illegal. Therefore, I would choose to ACK - BYE it.
From RFC 4028 section 7.2 - UAC Behaviour - Processing a 2xx Response:
If there was a Require header field in the response with the value
'timer', th
Hi All
Need inputs in the following scenario (wrt UAS)
- INVITE received with Session-Expires, Min-SE, Supported:timer, no refresher
is present
- 183 sent with Supported:timer
- UPDATE received with *no* Session-Expires, *no* Min-SE, Supported:timer, no
refresher is present
- 200 OK of UPDATE s
Hi All,
How to test a VoIP application through automation framework ?
regards
Satya
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