[asterisk-dev] [Code Review] PineTree:: Matching SIP peers beyond SIP proxy

[David Vossel]

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Other than the comments below, I'd clean up some of the excess debug information as well.


/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/348/#comment2529>

    This is really minor. It might be nice to have a comment with a VIA header example so it's easier to follow how its being parsed.



/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/348/#comment2530>

    This function looks pretty solid to me, as long as the ' ' is guaranteed to always be present between the transport and the beginning of the host.



/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/348/#comment2536>

    This is a very complicated for loop.  Here are a few ideas I have to make it at least appear safer and possibly avoid any weirdness that could occur.   Really, I'd just try and rethink this a little.
    
    1. The "if (ast_strlen_zero(viaheader))" block should explicitly return either TRUE or FALSE... I'd do this by making the internal if statments, (line <=) and (!findsecond), if, else if, and else statements, with the final else returning FALSE by default.
    
    2.  I'd make the "if (line == 2 && findsecond)" block an else if block right under the "if (ast_strlen_zero(viaheader))" block.  This keeps all the return statements together.



/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/348/#comment2538>

    Are we guaranteed previous is not NULL here?  What if the Via header is not present for some reason. 



/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/348/#comment2539>

    Does this need to be passed a transport type as well so the standard port can be set correctly if port is not present?


- David


On 2009-09-01 09:08:00, Olle E Johansson wrote:
> 
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> 
> (Updated 2009-09-01 09:08:00)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> 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
> -----
> 
>   /trunk/channels/chan_sip.c 215108 
>   /trunk/configs/sip.conf.sample 215108 
> 
> Diff: https://reviewboard.asterisk.org/r/348/diff
> 
> 
> Testing
> -------
> 
> Testing in private networks. Have had this code in production with customer for a couple of months, albeit on 1.4.
> 
> 
> Thanks,
> 
> Olle E
> 
>