[Asterisk-code-review] res pjsip: Add option to force G.726 to be treated as AAL2 p... (asterisk[13])

[Kevin Harwell]

Hello Matt Jordan, Joshua Colp,

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

    https://gerrit.asterisk.org/651

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

Change subject: res_pjsip: Add option to force G.726 to be treated as AAL2 packed.
......................................................................

res_pjsip: Add option to force G.726 to be treated as AAL2 packed.

Some phones send g.726 audio packed for AAL2, which differs from what is
recommended by RFC 3351. If Asterisk receives audio formatted as such when
negotiating g.726 then it sounds a bit distorted. Added an option to
res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726
AAL2 packed.

ASTERISK-25158 #close
Reported by: Steve Pitts

Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615
---
M CHANGES
M configs/samples/pjsip.conf.sample
A contrib/ast-db-manage/config/versions/28b8e71e541f_add_g726_non_standard.py
M include/asterisk/res_pjsip.h
M res/res_pjsip.c
M res/res_pjsip/pjsip_configuration.c
M res/res_pjsip_sdp_rtp.c
7 files changed, 64 insertions(+), 6 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/51/651/2
-- 
To view, visit https://gerrit.asterisk.org/651
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Gerrit-MessageType: newpatchset
Gerrit-Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>