[asterisk-dev] [Code Review] SIP: peer matching by callbackextension
Nick Lewis
Nick.Lewis at atltelecom.com
Tue Jun 15 08:14:09 CDT 2010
> On 2009-10-02 14:47:44, Olle E Johansson wrote:
> > I still think we should think very carefully and work on design before we commit either this or my peermatching stuff. I do not recommend merging this at this time.
> >
> > (As expressed earlier on the asterisk-dev mailinglist)
> > /O
>
> David Vossel wrote:
> If you don't have any immediate plans to work on a new peer matching algorithm how do you feel about this patch being committed? I know its not a pretty solution, but its simple and addresses the issue at hand. If a new algorithm is implemented in the future, I doubt this patch will add any complexity to it's integration.
I think 8 months is long enough to wait for comments on something as simple as matching the registered extension to the peer that registered it. May I request that you take Russell's comments and commit.
- Nick
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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/344/#review1139
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On 2009-10-08 16:39:29, David Vossel wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/344/
> -----------------------------------------------------------
>
> (Updated 2009-10-08 16:39:29)
>
>
> Review request for Asterisk Developers.
>
>
> Summary
> -------
>
> If there are a number of peers with different callbackextension parameters and the same host address. The first peer found matching the address is used regardless if that peer's callbackextension matches the incoming extension or not.
>
> Now, to better match peers with incoming calls, if an incoming call's address can match multiple peers by address, we check each of those peer's callbackextension against the incoming extension for the best possible match.
>
> It is possible that my implementation may be too expensive and only serve to address a minor edge case in the usage of chan_sip. I do not fully understand the impact my changes may have upon performance when a large number of peers are present. This patch assumes the new parse_uri() change has been made.
>
> -------------------------------------------
> for example with two peers as follows
> [trunk1]
> host=sip.myitsp.com
> callbackextension=9991
> ...
> [trunk2]
> host=sip.myitsp.com
> callbackextension=9992
> ...
>
> incoming calls to 9991 and to 9992 are both matched to the peer trunk1
> --------------------------------------------
>
>
> This addresses bug 14340.
> https://issues.asterisk.org/view.php?id=14340
>
>
> Diffs
> -----
>
> /trunk/channels/chan_sip.c 222947
> /trunk/configs/sip.conf.sample 222883
>
> Diff: https://reviewboard.asterisk.org/r/344/diff
>
>
> Testing
> -------
>
> Tested multiple peers with the same address containing different callbackextensions. Verified the correct peers were matched with incoming calls.
>
>
> Thanks,
>
> David
>
>
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