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

[Joshua Colp]

Attention is currently required from: Florentin Mayer.
Joshua Colp 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 2: Code-Review-1

(2 comments)

Patchset:

PS2: 
I think this should also have a test added.


File res/res_pjsip_sdp_rtp.c:

https://gerrit.asterisk.org/c/asterisk/+/17519/comment/fd35ac07_ca8af4af 
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 in packetization getting lost if ptime is present.



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Gerrit-Project: asterisk
Gerrit-Branch: 18
Gerrit-Change-Id: I3aabcfed3f379c36654f59c1872c313d0cb57e25
Gerrit-Change-Number: 17519
Gerrit-PatchSet: 2
Gerrit-Owner: Florentin Mayer <f.mayer at commend.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: Joshua Colp <jcolp at sangoma.com>
Gerrit-Attention: Florentin Mayer <f.mayer at commend.com>
Gerrit-Comment-Date: Fri, 03 Dec 2021 12:59:42 +0000
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