[asterisk-dev] [Code Review] Properly route responses according to the Via headers in the request

[Klaus Darilion]

Just in case somebody wants to read the relevant text in the SIP RFCs 
for response routing (there is no difference for response routing for 
proxies, SIP phones or PBX):

If the SIP node supports "rport", the relevant text is in section 4 of 
RFC 3581.
http://tools.ietf.org/html/rfc3581#section-4

This section also refers to section 18.2.2 of RFC 3261.
http://tools.ietf.org/html/rfc3261#section-18.2.2

regards
Klaus

On 16.12.2010 18:29, Matthew Nicholson wrote:
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1059/
>
>
> Review request for Asterisk Developers.
> By Matthew Nicholson.
>
>
>   Description
>
> This patch makes asterisk respect the Via headers in a request when responding to it. This is necessary in the even that a stateless proxy is in between asterisk and the requester. Without this patch, the response is simply routed back to the address we received the initial request from.
>
>
>   Testing
>
> Briefly tested using openser as a stateless proxy and another asterisk machine as the requester.  I sent and invite, then some INFO DTMF messages.  Without the patch, our asterisk machine sends all responses to the INFO requests to the proxy, with the patch they are properly routed to the requesting asterisk machine.  I also briefly tested with openser configured to use the Record-Route header as a stateful proxy.
>
>
>   Diffs
>
>     * /branches/1.4/channels/chan_sip.c (294163)
>
> View Diff <https://reviewboard.asterisk.org/r/1059/diff/>
>
>
>
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