[asterisk-dev] [Code Review] 3244: res_pjsip_sdp_rtp: Apply packetization rules on inbound SDP handling
Joshua Colp
reviewboard at asterisk.org
Fri Feb 21 10:04:21 CST 2014
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3244/#review10921
-----------------------------------------------------------
Ship it!
Ship It!
- Joshua Colp
On Feb. 20, 2014, 10:29 p.m., Matt Jordan wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3244/
> -----------------------------------------------------------
>
> (Updated Feb. 20, 2014, 10:29 p.m.)
>
>
> Review request for Asterisk Developers, Joshua Colp and Mark Michelson.
>
>
> Repository: Asterisk
>
>
> Description
> -------
>
> The setting 'use_ptime' is supposed to tell Asterisk to honour the ptime attribute in an offer, preferring it to whatever packetization preferences have been set internally. Currently, however, something rather quirky will happen:
>
> (1) The SDP answer will be constructed in create_outgoing_sdp_stream. This will use the preferences from the endpoint, such that the 200 OK response will add the packetization preferences from the endpoint, and not what was offered.
> (2) When the 200 response is issued, apply_negotiated_sdp_stream is called. This will call apply_packetization, which will use the ptime attribute from the offer internally.
>
> We end up telling the offerer to use the internal ptime attribute, but we end up using the offered ptime attribute. Hilarity ensues.
>
> This patch modifies the behaviour by calling apply_packetization from negotiate_incoming_sdp_stream, which is called prior to create_outgoing_sdp_stream. This causes the format preferences on the session's media object to be set to the inbound ptime value (if 'use_ptime' is enabled), such that the construction of the answer gets the right value immediately.
>
>
> Diffs
> -----
>
> /branches/12/res/res_pjsip_sdp_rtp.c 408501
>
> Diff: https://reviewboard.asterisk.org/r/3244/diff/
>
>
> Testing
> -------
>
> The packetization test suite test that verifies 'use_ptime' now passes. Tests that cover 'use_ptime' being set to False continue to pass.
>
> These tests will be put up for a review when more of them are done.
>
>
> Thanks,
>
> Matt Jordan
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140221/0e608f11/attachment.html>
More information about the asterisk-dev
mailing list