[asterisk-dev] Asterisk 1.8 SIP_CAUSE performance regression

[Bryant Zimmerman]

----------------------------------------

From: "ik" <idokan at gmail.com>

Sent: Thursday, August 18, 2011 9:47 AM

To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>

Subject: Re: [asterisk-dev] Asterisk 1.8 SIP_CAUSE performance regression


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

>

>

>

Does it give the same data as the hangupcause when you use the h context or 
is it different?


Bryant


-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20110818/ce766cc8/attachment.htm>