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

[Olle E. Johansson]

3 sep 2010 kl. 15.58 skrev Klaus Darilion:

> Hi Olle!
> 
> What's the status of this patch?
It lies happily somewhere and needs love. The branch needs to be resolved, as there are commit conflicts in it. And someone must code support for parsing the received header, as you suggested. 

/O
> 
> thanks
> Klaus
> 
> 
> 12 jul 2010 kl. 16.18 skrev David Vossel:
> 
>> 
>> -----------------------------------------------------------
>> This is an automatically generated e-mail. To reply, visit:
>> https://reviewboard.asterisk.org/r/348/#review2362
>> -----------------------------------------------------------
>> 
>> 
>> 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.
> Hmm. Something strange must have happened while producing a diff, 
> because that's not really my intention... ;-)
> 
> Will try again.
> 
> /O
> 
> 
>> 
>> - David
>> 
>> 
>> On 2010-07-09 08:16:07, Olle E Johansson wrote:
>>> 
>>> -----------------------------------------------------------
>>> This is an automatically generated e-mail. To reply, visit:
>>> https://reviewboard.asterisk.org/r/348/
>>> -----------------------------------------------------------
>>> 
>>> (Updated 2010-07-09 08:16:07)
>>> 
>>> 
>>> 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 274865
>>> 
>>> 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
>>> 
>>> 
> 
> ---
> * Olle E Johansson - oej at edvina.net
> * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
> 
> 
> 
> 
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---
* Olle E Johansson - oej at edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden