[asterisk-commits] tests/channels/pjsip/sdp offer answer: Update Opus rtpmap. (testsuite[master])

[SVN commits to the Asterisk project]

Anonymous Coward #1000019 has submitted this change and it was merged.

Change subject: tests/channels/pjsip/sdp_offer_answer: Update Opus rtpmap.
......................................................................


tests/channels/pjsip/sdp_offer_answer: Update Opus rtpmap.

The Opus codec is negotiated with 2 channels. This has been fixed in
Asterisk but the test needs to be updated to use it for negotiation.

ASTERISK-24779

Change-Id: I34d45aa753e8024f9797b6dc18d0524e5f112303
---
M tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/basic/sipp/uac-all-codecs.xml
1 file changed, 1 insertion(+), 1 deletion(-)

Approvals:
  Anonymous Coward #1000019: Verified
  Matt Jordan: Looks good to me, approved



diff --git a/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/basic/sipp/uac-all-codecs.xml b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/basic/sipp/uac-all-codecs.xml
index 84ba2aa..fa41aed 100644
--- a/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/basic/sipp/uac-all-codecs.xml
+++ b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/basic/sipp/uac-all-codecs.xml
@@ -42,7 +42,7 @@
       a=rtpmap:102 G7221/16000
       a=rtpmap:115 G7221/32000
       a=rtpmap:116 G719/48000
-      a=rtpmap:107 opus/48000
+      a=rtpmap:107 opus/48000/2
       a=rtpmap:101 telephone-event/8000
       a=fmtp:101 0-16
       a=ptime:20

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Gerrit-MessageType: merged
Gerrit-Change-Id: I34d45aa753e8024f9797b6dc18d0524e5f112303
Gerrit-PatchSet: 1
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>