[asterisk-dev] [Code Review] PineTree:: Matching SIP peers beyond SIP proxy
David Vossel
dvossel at digium.com
Mon Jul 12 09:18:00 CDT 2010
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With 1481 changes to chan_sip this is very difficult to review. I'd understand if those were all functional changes, but 99% of what I saw looked like it was reverting things to an older version of trunk. I don't think we want to revert some of the changes we made involving the splitting up of chan_sip.c into separate files. I believe this patch just needs to be updated to a current version of trunk. Otherwise it is very difficult to find your changes in all that noise.
- David
On 2010-07-09 08:16:07, Olle E Johansson wrote:
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> https://reviewboard.asterisk.org/r/348/
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> (Updated 2010-07-09 08:16:07)
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>
> Review request for Asterisk Developers.
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>
> Summary
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>
> Many of us implement asterisk behind SIP proxys for load balancing or
> failover or both. That means that all messages to Asterisk is sent by
> the proxy and all peer matching on IP/port fails. Asterisk simply
> doesn't know how to separate the devices behind the proxy.
>
> With my new code, you can add a rule to the SIP proxy [peer] section,
> saying "don't match me, match who sent to me". The way Asterisk does
> that, is by reading the second VIA header. This is the device that
> sent the message to Asterisk - another proxy or an endpoint. You can
> also be very strict and say "match last via" - which always will be
> the other endpoint.
>
> The benefit of all this is that all Asterisk features now work -
> accountcode, codec settings, authentication. You can provision
> different SIP trunks with different features, even though Asterisk is
> hidden by a proxy.
>
> For outbound calls, you use the outbound proxy setting as before.
>
>
> Diffs
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> /trunk/channels/chan_sip.c 274865
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> Diff: https://reviewboard.asterisk.org/r/348/diff
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>
> Testing
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> Testing in private networks. Have had this code in production with customer for a couple of months, albeit on 1.4.
>
>
> Thanks,
>
> Olle E
>
>
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