[asterisk-dev] [Code Review] 2493: Add WebSocket transport module

[Jason Parker]

> On May 9, 2013, 5:12 p.m., Joshua Colp wrote:
> > /team/group/pimp_my_sip/res/res_sip_transport_websocket.c, line 291
> > <https://reviewboard.asterisk.org/r/2493/diff/2/?file=37424#file37424line291>
> >
> >     This now introduces more of a bottle neck on outgoing requests and I think using the URI isn't safe (since it's possible an implementation may use the same contact across multiple instances, even though it probably shouldn't).
> >     
> >     Doesn't pjsip have a mechanism to do resolution to active transports based on the URI already? Is it possible to leverage that and add a parameter to the URI to make it unique?
> >     
> >     Regardless to make this more lightweight you could examine the transport on the URI and see if it is for websocket, if not then don't even do a lookup.
> 
> Jason Parker wrote:
>     Finding a transport requires DNS resolution.  We can't let it try to do that, because of the .invalid TLD that gets used.
>     
>     I'm not finishing this thought right now, since reviewboard currently hates me.

...something, something, DNS failure.  I did your optimization.


- Jason


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On May 8, 2013, 4:13 p.m., Jason Parker wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2493/
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> 
> (Updated May 8, 2013, 4:13 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-20952
>     https://issues.asterisk.org/jira/browse/ASTERISK-20952
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> Adds a custom WebSocket transport module.
> 
> 
> Diffs
> -----
> 
>   /team/group/pimp_my_sip/include/asterisk/res_sip.h 387967 
>   /team/group/pimp_my_sip/res/res_sip.c 387967 
>   /team/group/pimp_my_sip/res/res_sip.exports.in 387967 
>   /team/group/pimp_my_sip/res/res_sip/config_transport.c 387967 
>   /team/group/pimp_my_sip/res/res_sip/location.c 387967 
>   /team/group/pimp_my_sip/res/res_sip_outbound_registration.c 387967 
>   /team/group/pimp_my_sip/res/res_sip_transport_websocket.c PRE-CREATION 
> 
> Diff: https://reviewboard.asterisk.org/r/2493/diff/
> 
> 
> Testing
> -------
> 
> Registration works, incoming (from the browser) and outgoing (to the browser) calls work, audio flows (though, I don't have a mic, so I didn't test audio that direction).
> 
> 
> Thanks,
> 
> Jason Parker
> 
>

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