[Asterisk-code-review] chan_sip: Hold/Resume an sRTP call which had other/less media streams. (asterisk[16])

Joshua Colp asteriskteam at digium.com
Wed Jan 6 12:03:08 CST 2021


Attention is currently required from: Alexander Traud.
Joshua Colp has posted comments on this change. ( https://gerrit.asterisk.org/c/asterisk/+/15272 )

Change subject: chan_sip: Hold/Resume an sRTP call which had other/less media streams.
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Patch Set 1: Code-Review-1

(1 comment)

Patchset:

PS1: 
It appears this change introduces a behavior change or regression as uncovered by the tests/channels/SIP/SDP_offer_answer test. In that test the decline_crypto.xml scenario file is receiving a different SIP response than it did previously. The scenario file expects to receive a 488 SIP response while it is now receiving a 603. In the specific case I believe it should rightfully receive a 488.



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Gerrit-Project: asterisk
Gerrit-Branch: 16
Gerrit-Change-Id: Ie49e4e2a11f0265f914b684738348ba8c0f89755
Gerrit-Change-Number: 15272
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Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
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Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
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