[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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> 
> (Updated 2010-07-09 08:16:07)
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> 
> Review request for Asterisk Developers.
> 
> 
> 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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