[Asterisk-code-review] res_pjsip_sdp_rtp: Preserve order of RTP codecs (asterisk[18])

Florentin Mayer asteriskteam at digium.com
Mon Dec 13 08:26:12 CST 2021


Attention is currently required from: Joshua Colp.
Florentin Mayer has posted comments on this change. ( https://gerrit.asterisk.org/c/asterisk/+/17519 )

Change subject: res_pjsip_sdp_rtp: Preserve order of RTP codecs
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Patch Set 3:

(2 comments)

Patchset:

PS2: 
> I think this should also have a test added.
As this is my first commit in the Asterisk universe, I'm not so sure if I've added my test at the correct place.

As the commit changes the SDP to ast_format parsing, I can't add a unit/integration test in the asterisk repo?. I now added the test to "channels/pjsip/hold"  as this was the operation where the initial bug occurred. If this is not what you have expected please let me know.


File res/res_pjsip_sdp_rtp.c:

https://gerrit.asterisk.org/c/asterisk/+/17519/comment/ba8df652_99249155 
PS2, Line 333: 	ast_format_cap_remove_by_type(astformats, AST_MEDIA_TYPE_UNKNOWN);
> The framing doesn't appear to be getting set on astformats at the bottom of this function, resulting […]
Done



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Gerrit-Project: asterisk
Gerrit-Branch: 18
Gerrit-Change-Id: I3aabcfed3f379c36654f59c1872c313d0cb57e25
Gerrit-Change-Number: 17519
Gerrit-PatchSet: 3
Gerrit-Owner: Florentin Mayer <f.mayer at commend.com>
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Gerrit-Reviewer: Joshua Colp <jcolp at sangoma.com>
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