[asterisk-dev] [Code Review] 2445: Pimp my SIP: Media Negotiations
Joshua Colp
reviewboard at asterisk.org
Tue Apr 16 17:43:18 CDT 2013
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/team/group/pimp_my_sip/channels/chan_gulp.c
<https://reviewboard.asterisk.org/r/2445/#comment15964>
I think this approach is overkill, a lot. You've already got the req_caps format capabilities for the requested capabilities - a dialplan function which overwrites it for the specific media type would do the exact same job. You could also extend usage of it so it is always populated and allow it to be queried using the same dialplan function.
/team/group/pimp_my_sip/include/asterisk/res_sip_session.h
<https://reviewboard.asterisk.org/r/2445/#comment15965>
This description is incorrect. You are sending an INVITE/starting a session.
- Joshua Colp
On April 11, 2013, 10:18 p.m., Kevin Harwell wrote:
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> https://reviewboard.asterisk.org/r/2445/
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> (Updated April 11, 2013, 10:18 p.m.)
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>
> Review request for Asterisk Developers.
>
>
> Bugs: ASTERISK-21186
> https://issues.asterisk.org/jira/browse/ASTERISK-21186
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> Repository: Asterisk
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> Description
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> Added a dialplan function MEDIA_OFFER that accepts a codec type (example: 'audio') and allows overriding, or re-ordering, of an endpoints codecs prior to dialing (e.g. using a pre-dial handler). This adds functionality for outbound requests only.
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> Example: Set(MEDIA_OFFER(audio)=ulaw,g722) ; sets the outgoing codecs to be ulaw,g722
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> Note that using this function and setting new media offers completely overrides what is specified on the endpoint. Currently it is allowed to even list a codec that was not previously specified on the endpoint.
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> The code allows for un/registering of media offer types that can be associated with the function itself. This allows for future expansion of other types, for example T.38. Types 'audio' and 'video' are currently supported.
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> Diffs
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> /team/group/pimp_my_sip/res/res_sip_session.exports.in 385384
> /team/group/pimp_my_sip/res/res_sip_session.c 385384
> /team/group/pimp_my_sip/channels/chan_gulp.c 385384
> /team/group/pimp_my_sip/res/res_sip_sdp_rtp.c 385384
> /team/group/pimp_my_sip/include/asterisk/res_sip_session.h 385384
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> Diff: https://reviewboard.asterisk.org/r/2445/diff/
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>
> Testing
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> Ran through several scenarios setting new MEDIA_OFFER(s). Tested re-ordering of already specified codecs on an endpoint, tested setting only a single codec (both specified and not on endpoint). Tested reading back out the newly set codecs in the dialplan.
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> Thanks,
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> Kevin Harwell
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>
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