[asterisk-commits] bundled multi-stream test: Add missing a=ssrc attributes. (testsuite[master])

[SVN commits to the Asterisk project]

Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/6544 )

Change subject: bundled multi-stream test: Add missing a=ssrc attributes.
......................................................................

bundled multi-stream test: Add missing a=ssrc attributes.

Change-Id: I1f25d7eff8ca2f754e8adc53de07c2bb906b1f12
---
M tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/sipp/uac-multiple-video-with-audio.xml
1 file changed, 8 insertions(+), 6 deletions(-)

Approvals:
  Joshua Colp: Looks good to me, but someone else must approve
  Matthew Fredrickson: Looks good to me, approved
  Jenkins2: Approved for Submit



diff --git a/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/sipp/uac-multiple-video-with-audio.xml b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/sipp/uac-multiple-video-with-audio.xml
index 4d52ad8..cd6d2e7 100644
--- a/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/sipp/uac-multiple-video-with-audio.xml
+++ b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/sipp/uac-multiple-video-with-audio.xml
@@ -4,7 +4,6 @@
 <scenario name="Basic Sipstone UAC">
   <send retrans="500">
     <![CDATA[
-
       INVITE sip:answer@[remote_ip]:[remote_port] SIP/2.0
       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
       From: alice <sip:[service]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
@@ -33,21 +32,28 @@
       a=maxptime:20
       a=sendrecv
       a=mid:audio
+      a=rtcp-mux
+      a=ssrc:1 cname:alice
       m=video 6001 RTP/AVP 99 34
       a=rtpmap:99 H264/90000
       a=rtpmap:34 H263/90000
       a=sendrecv
       a=mid:video
+      a=rtcp-mux
+      a=ssrc:2 cname:bob
       m=video 6002 RTP/AVP 99
       c=IN IP[media_ip_type] [media_ip]
       a=rtpmap:99 H264/90000
       a=sendrecv
       a=mid:video
+      a=rtcp-mux
+      a=ssrc:3 cname:charlie
       m=video 6003 RTP/AVP 34
       a=rtpmap:34 H263/90000
       a=sendrecv
       a=mid:video
-
+      a=rtcp-mux
+      a=ssrc:4 cname:david
     ]]>
   </send>
 
@@ -79,7 +85,6 @@
 
   <send>
     <![CDATA[
-
       ACK sip:answer@[remote_ip]:[remote_port] SIP/2.0
       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
       From: alice <sip:[service]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
@@ -90,7 +95,6 @@
       Max-Forwards: 70
       Subject: Codec Test
       Content-Length: 0
-
     ]]>
   </send>
 
@@ -99,7 +103,6 @@
 
   <send>
     <![CDATA[
-
       SIP/2.0 200 OK
       [last_Via:]
       [last_From:]
@@ -108,7 +111,6 @@
       [last_CSeq:]
       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
       Content-Length: 0
-
     ]]>
   </send>
 

-- 
To view, visit https://gerrit.asterisk.org/6544
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Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-MessageType: merged
Gerrit-Change-Id: I1f25d7eff8ca2f754e8adc53de07c2bb906b1f12
Gerrit-Change-Number: 6544
Gerrit-PatchSet: 1
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Matthew Fredrickson <creslin at digium.com>
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