[asterisk-commits] rtp engine: allocate RTP dynamic payloads per session (testsuite[master])

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Mar 24 17:47:30 CDT 2017


Anonymous Coward #1000019 has submitted this change and it was merged. ( https://gerrit.asterisk.org/5261 )

Change subject: rtp_engine: allocate RTP dynamic payloads per session
......................................................................


rtp_engine: allocate RTP dynamic payloads per session

Dynamic RTP payload types can now be truly dynamically allocated instead of
being statically defined. When possible they will still default to the pre-
designated value Asterisk used to statically define them. This caused a few
tests to fail that were matching against had hard-coded payload types values.
Allowing Asterisk to default the dyanamic payload types alleviated the test
failures, however, this patch goes ahead and subsitutes a regex pattern match
for those failing tests just in case.

ASTERISK-26515 #close

Change-Id: I9070e13981c1c43cc09a7640bc93563df3f18263
---
M tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264.xml
M tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264_13.xml
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_h264.xml
M tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/basic/sipp/uac-basic-codecs-delayed.xml
M tests/channels/pjsip/sdp_offer_answer/incoming/nominal/single-media-stream/video/basic/sipp/uac-basic-codecs-delayed.xml
6 files changed, 9 insertions(+), 5 deletions(-)

Approvals:
  Mark Michelson: Looks good to me, but someone else must approve
  Anonymous Coward #1000019: Verified
  Joshua Colp: Looks good to me, approved
  Matthew Fredrickson: Looks good to me, but someone else must approve



diff --git a/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264.xml b/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264.xml
index 593da42..1716969 100644
--- a/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264.xml
+++ b/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264.xml
@@ -11,7 +11,7 @@
 				search_in="hdr"
 				check_it="true"
 				assign_to="global_call_id"/>
-			<ereg regexp="m=video [0-9]{1,5} RTP/AVP( [0-9]{1,3})+..*a=rtpmap:99 H264/90000.*a=fmtp:99 profile-level-id=42801E;max-mbps=48600;packetization-mode=1"
+			<ereg regexp="m=video [0-9]{1,5} RTP/AVP( [0-9]{1,3})+..*a=rtpmap:9[6-9]|1{0,1}[0-9]|12[0-7] H264/90000.*a=fmtp:9[6-9]|1{0,1}[0-9]|12[0-7] profile-level-id=42801E;max-mbps=48600;packetization-mode=1"
 			      search_in="body" check_it="true" assign_to="1"/>
 			<strcmp assign_to="1" variable="1" value=""/>
 
diff --git a/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264_13.xml b/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264_13.xml
index 88d1809..0efece2 100644
--- a/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264_13.xml
+++ b/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264_13.xml
@@ -11,7 +11,7 @@
 				search_in="hdr"
 				check_it="true"
 				assign_to="global_call_id"/>
-			<ereg regexp="m=video [0-9]{1,5} RTP/AVP( [0-9]{1,3})+..*a=rtpmap:99 H264/90000.*a=fmtp:99 max-mbps=48600;packetization-mode=1;profile-level-id=42801E"
+			<ereg regexp="m=video [0-9]{1,5} RTP/AVP( [0-9]{1,3})+..*a=rtpmap:9[6-9]|1{0,1}[0-9]|12[0-7] H264/90000.*a=fmtp:9[6-9]|1{0,1}[0-9]|12[0-7] max-mbps=48600;packetization-mode=1;profile-level-id=42801E"
 			      search_in="body" check_it="true" assign_to="1"/>
 			<strcmp assign_to="1" variable="1" value=""/>
 
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 b9a553e..f4f304f 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
@@ -1,3 +1,7 @@
+[global]
+type=global
+debug=yes
+
 [local-transport-udp]
 type=transport
 bind=127.0.0.1
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 88d1809..0efece2 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
@@ -11,7 +11,7 @@
 				search_in="hdr"
 				check_it="true"
 				assign_to="global_call_id"/>
-			<ereg regexp="m=video [0-9]{1,5} RTP/AVP( [0-9]{1,3})+..*a=rtpmap:99 H264/90000.*a=fmtp:99 max-mbps=48600;packetization-mode=1;profile-level-id=42801E"
+			<ereg regexp="m=video [0-9]{1,5} RTP/AVP( [0-9]{1,3})+..*a=rtpmap:9[6-9]|1{0,1}[0-9]|12[0-7] H264/90000.*a=fmtp:9[6-9]|1{0,1}[0-9]|12[0-7] max-mbps=48600;packetization-mode=1;profile-level-id=42801E"
 			      search_in="body" check_it="true" assign_to="1"/>
 			<strcmp assign_to="1" variable="1" value=""/>
 
diff --git a/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/basic/sipp/uac-basic-codecs-delayed.xml b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/basic/sipp/uac-basic-codecs-delayed.xml
index c7c86ca..2c26370 100644
--- a/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/basic/sipp/uac-basic-codecs-delayed.xml
+++ b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/basic/sipp/uac-basic-codecs-delayed.xml
@@ -37,7 +37,7 @@
       <ereg regexp="m=audio [0-9]{1,5} RTP/AVP 9 0 8 101+..*"
             search_in="body" check_it="true" assign_to="1"/>
       <test assign_to="1" variable="1" compare="equal" value=""/>
-      <ereg regexp="m=video [0-9]{1,5} RTP/AVP 99 34+..*"
+      <ereg regexp="m=video [0-9]{1,5} RTP/AVP 9[6-9]|1{0,1}[0-9]|12[0-7] 34+..*"
             search_in="body" check_it="true" assign_to="2"/>
       <test assign_to="2" variable="2" compare="equal" value=""/>
     </action>
diff --git a/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/single-media-stream/video/basic/sipp/uac-basic-codecs-delayed.xml b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/single-media-stream/video/basic/sipp/uac-basic-codecs-delayed.xml
index 33f9975..bac64bf 100644
--- a/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/single-media-stream/video/basic/sipp/uac-basic-codecs-delayed.xml
+++ b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/single-media-stream/video/basic/sipp/uac-basic-codecs-delayed.xml
@@ -34,7 +34,7 @@
 
   <recv response="200" rtd="true">
     <action>
-      <ereg regexp="m=video [0-9]{1,5} RTP/AVP 99 34+..*"
+      <ereg regexp="m=video [0-9]{1,5} RTP/AVP 9[6-9]|1{0,1}[0-9]|12[0-7] 34+..*"
             search_in="body" check_it="true" assign_to="1"/>
       <test assign_to="1" variable="1" compare="equal" value=""/>
     </action>

-- 
To view, visit https://gerrit.asterisk.org/5261
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Gerrit-MessageType: merged
Gerrit-Change-Id: I9070e13981c1c43cc09a7640bc93563df3f18263
Gerrit-PatchSet: 1
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Matthew Fredrickson <creslin at digium.com>



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