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

[Florentin Mayer]

Hello Friendly Automation, 

I'd like you to reexamine a change. Please visit

    https://gerrit.asterisk.org/c/asterisk/+/17519

to look at the new patch set (#2).

Change subject: res_pjsip_sdp_rtp: Preserve order of RTP codecs
......................................................................

res_pjsip_sdp_rtp: Preserve order of RTP codecs

The ast_rtp_codecs_payloads functions do not preserve the order in which
the payloads were specified on an incoming SDP media line. This leads to
a problem with the codec negotiation functionality, as the format
capabilities of the stream are extracted from the ast_rtp_codecs. This
commit moves the ast_rtp_codec to ast_format conversion to the place
where the order is still known.

ASTERISK-28863
ASTERISK-29320

Change-Id: I3aabcfed3f379c36654f59c1872c313d0cb57e25
---
M res/res_pjsip_sdp_rtp.c
1 file changed, 19 insertions(+), 10 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/19/17519/2
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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-MessageType: newpatchset
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