[Asterisk-code-review] res pjsip sdp rtp: Control endpoints media address use per c... (asterisk[13])

[Salah Ahmed]

Salah Ahmed has posted comments on this change. ( https://gerrit.asterisk.org/10431 )

Change subject: res_pjsip_sdp_rtp: Control endpoints media_address use per call basis
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Patch Set 7:

> (2 comments)
 > 
 > This is a new feature so it should also have a test to confirm that
 > the dialplan variable does as it should and doesn't break in the
 > future. If no test is provided then it would be eligible for
 > master.

We wrote several testcases and extensively test this patch, this patch does not work for inbound endpoints. If we configure a media_address for an inbound  

 > (2 comments)
 > 
 > This is a new feature so it should also have a test to confirm that
 > the dialplan variable does as it should and doesn't break in the
 > future. If no test is provided then it would be eligible for
 > master.

We wrote some test cases and run few extensive tests on this patch. This patch break if we configure media_address for an inbound endpoint. We tried to solve this but this would be complex to implement. So far we understand from code reading, the conn information of A-leg's SDP is generated while an INVITE received and this pre-generated SDP is used while the answer message sends to A side. We can alter the connection IP in answer SDP, but could not alter the origin IP.


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Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-MessageType: comment
Gerrit-Change-Id: I852adbee9f8b034ec332fbe4e1b6692cb2939518
Gerrit-Change-Number: 10431
Gerrit-PatchSet: 7
Gerrit-Owner: Salah Ahmed <txrubel at gmail.com>
Gerrit-Reviewer: Corey Farrell <git at cfware.com>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2 (1000185)
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Salah Ahmed <txrubel at gmail.com>
Gerrit-Reviewer: Sean Bright <sean.bright at gmail.com>
Gerrit-Comment-Date: Fri, 19 Oct 2018 20:40:29 +0000
Gerrit-HasComments: No
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