[asterisk-commits] pjsip: Correct attribute passthrough tests. (testsuite[master])

[SVN commits to the Asterisk project]

Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/5879 )

Change subject: pjsip: Correct attribute passthrough tests.
......................................................................

pjsip: Correct attribute passthrough tests.

While testing multistream support in PJSIP it uncovered
a few issues with the attribute passthrough tests.

The speex test was not configured to accept both ulaw
and speex. As a result in multistream a reinvite occurred
for the calling leg to switch it to only speex. This change
makes both ulaw and speex accepted so that no reinvite occurs.

The h263 and h264 tests did not answer their INVITEs with
valid SDP. They answered with only a single video stream
when they should answer with both an audio and video. In
multistream this caused the calling leg to get a reinvite
declining the audio stream. PJSIP helps in this area by
correcting the received SDP so it is proper.

Change-Id: I966e7c53831e0ba925e3c8fc500953b9c46a4b1b
---
M tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/pjsip.conf
M tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_B_h263.xml
M tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_B_h264.xml
3 files changed, 5 insertions(+), 0 deletions(-)

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



diff --git a/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/pjsip.conf b/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/pjsip.conf
index f4f304f..6b9089d 100644
--- a/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/pjsip.conf
+++ b/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/pjsip.conf
@@ -45,6 +45,7 @@
 type=peer
 aors=phoneB-speex
 disallow=all
+allow=ulaw
 allow=speex
 
 [phoneB-speex]
diff --git a/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_B_h263.xml b/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_B_h263.xml
index a132490..311d1f7 100644
--- a/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_B_h263.xml
+++ b/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_B_h263.xml
@@ -56,6 +56,8 @@
 			s=-
 			c=IN IP[media_ip_type] [media_ip]
 			t=0 0
+			m=audio 6000 RTP/AVP 0
+			a=rtpmap:0 PCMU/8000
 			m=video 6002 RTP/AVP 34
 			a=rtpmap:34 H263/90000
 			a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;CIF16=1;F=1;I=1;J=1;T=1;K=1;N=1;PAR=255:255;BPP=65535;HRD=1
diff --git a/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_B_h264.xml b/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_B_h264.xml
index 0efece2..5af8629 100644
--- a/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_B_h264.xml
+++ b/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_B_h264.xml
@@ -56,6 +56,8 @@
 			s=-
 			c=IN IP[media_ip_type] [media_ip]
 			t=0 0
+			m=audio 6000 RTP/AVP 0
+			a=rtpmap:0 PCMU/8000
 			m=video 6002 RTP/AVP 99
 			a=rtpmap:99 H264/90000
 			a=fmtp:99 profile-level-id=42801e;packetization-mode=1;max-mbps=48600

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Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-MessageType: merged
Gerrit-Change-Id: I966e7c53831e0ba925e3c8fc500953b9c46a4b1b
Gerrit-Change-Number: 5879
Gerrit-PatchSet: 1
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Matthew Fredrickson <creslin at digium.com>
Gerrit-Reviewer: Sean Bright <sean.bright at gmail.com>
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