[asterisk-dev] Asterisk 1.8 SIP_CAUSE performance regression

[ik]

On Thu, Aug 18, 2011 at 16:31, Matthew Nicholson <mnicholson at digium.com>wrote:

> When chan_sip sets MASTER_CHANNEL(HASH(SIP_CAUSE,<chan name>)) the
> MASTER_CHANNEL function scans the channel list which can be a slow
> operation. Under moderate traffic loads this causes delays in processing
> sip messages which can cause retransmission timers to expire.
>
> There is no other way to know the response code of a SIP message, but
> depending on what you are using the response codes for, you may be able
> to accomplish the same thing a different way. What are you using the
> codes for?
>

I write application using Asterisk, and I use it to better understand the
hangup cause, or error messages from SIP trunks.


Ido

>
> Also be aware that this feature is not going away, it simply will be
> disabled by default.
>
> On Thu, 2011-08-18 at 15:49 +0300, ik wrote:
> > I'm using it.
> >
> > Can you please provide more information on the issue with this
> > feature ?
> > Is there another way to know the response code of SIP ?
> >
> > Thanks,
> >
> > Ido
> >
> > On Thu, Aug 18, 2011 at 15:42, Matthew Nicholson
> > <mnicholson at digium.com> wrote:
> >         Greetings,
> >
> >         Recently a performance regression in chan_sip was discovered
> >         in Asterisk
> >         1.8. The regression is caused by chan_sip setting
> >         MASTER_CHANNEL(HASH(SIP_CAUSE,<chan name>)) after each
> >         response received
> >         on a channel. That feature has been made optional in the
> >         latest 1.8 SVN
> >         code, but is currently still enabled by default. After some
> >         internal
> >         discussion, we decided to consider disabling this feature by
> >         default in
> >         future 1.8 versions. This would be an unexpected behavior
> >         change for
> >         anyone depending on that SIP_CAUSE update in their dialplan.
> >         Alternatively, with this feature enabled, anyone upgrading
> >         from Asterisk
> >         1.4 will see a 60% decrease in the amount of SIP traffic they
> >         can handle
> >         before encountering problems.
> >
> >         Before disabling this feature, we wanted to get a feel for how
> >         many
> >         people are using it. If you use this feature, please respond
> >         to this
> >         email and let us know.
> >         --
> >         Matthew Nicholson
> >         Digium, Inc. | Software Developer
> >
> >
> >
> >
> >         --
> >
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> --
> Matthew Nicholson
> Digium, Inc. | Software Developer
>
>
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