[Asterisk-code-review] res pjsip session: outgoing call did not offer all configure... (asterisk[15])

[Kevin Harwell]

Hello Richard Mudgett, Jenkins2, Joshua Colp, 

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

    https://gerrit.asterisk.org/6569

to look at the new patch set (#3).

Change subject: res_pjsip_session: outgoing call did not offer all configured codecs
......................................................................

res_pjsip_session: outgoing call did not offer all configured codecs

For some scenarios when an outgoing call was made only a subset of the
configured codecs were offered. If the codecs being offered happened to
not have a codec supported by the phone then the call would fail.

For instance Alice and Bob both are configured in Asterisk for g722 and ulaw(
allow=!all,g722,ulaw). Alice's endpoint however only supports g722 while Bob's
only supports ulaw. When Alice calls Bob, Alice negotiates g722 fine with
Asterisk. But when Asterisk sends the outgoing offer to Bob it only contains
g722 and not both g722 and ulaw, so the call ends.

This patch makes it so all the audio codecs configured on the endpoint always
get sent, and not just a subset. However priority is given to those codecs that
are compatible with the "other side".

ASTERISK-27259 #close

Change-Id: Iffabc373bd94cd1dc700925dcfe406e12918c696
---
M res/res_pjsip_session.c
1 file changed, 10 insertions(+), 0 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/69/6569/3
-- 
To view, visit https://gerrit.asterisk.org/6569
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Gerrit-Project: asterisk
Gerrit-Branch: 15
Gerrit-MessageType: newpatchset
Gerrit-Change-Id: Iffabc373bd94cd1dc700925dcfe406e12918c696
Gerrit-Change-Number: 6569
Gerrit-PatchSet: 3
Gerrit-Owner: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
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