[Asterisk-code-review] pjsip sdp rtp: Add option endpoint/bind rtp to media address (asterisk[13])

[George Joseph]

Hello Anonymous Coward #1000019,

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

    https://gerrit.asterisk.org/1928

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

Change subject: pjsip_sdp_rtp:  Add option endpoint/bind_rtp_to_media_address
......................................................................

pjsip_sdp_rtp:  Add option endpoint/bind_rtp_to_media_address

On a system with multiple ip addresses in the same subnet, if a
transport is bound to a specific ip address and endpoint/media_address
 is set, the SIP/SDP will have the correct address in all fields but
the rtp stream MAY still originate from one of the other ip addresses,
most probably the "primary" ip address.  This happens because
 res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with
the "all" ip address (0.0.0.0 or ::).

The new option causes res_pjsip_sdp_rtp/create_rtp to call
ast_rtp_instance_new with the endpoint's media_address (if specified)
instead of the "all" address.  This causes the packets to originate from
the specified address.

ASTERISK-25632
ASTERISK-25637
Reported-by: Olivier Krief
Reported-by: Dan Journo

Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88
---
M CHANGES
M configs/samples/pjsip.conf.sample
A contrib/ast-db-manage/config/versions/26d7f3bf0fa5_add_bind_rtp_to_media_address_to_pjsip.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, 61 insertions(+), 1 deletion(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/28/1928/2
-- 
To view, visit https://gerrit.asterisk.org/1928
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Gerrit-MessageType: newpatchset
Gerrit-Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: George Joseph <george.joseph at fairview5.com>
Gerrit-Reviewer: Anonymous Coward #1000019