[Asterisk-code-review] pjsip channel: Use specified media port (testsuite[master])

[Kevin Harwell]

Kevin Harwell has uploaded this change for review. ( https://gerrit.asterisk.org/8860


Change subject: pjsip_channel: Use specified media port
......................................................................

pjsip_channel: Use specified media port

This test took advantage of the fact that SIPp defaults the media port to 6000,
which it then checked against for the test. However, a recent patch made it so
tests using the SIPp 'media_port' option would get an available random port
instead.

This patch no longer uses the 'media_port' option, but instead uses a hard
coded value for the media port in the SDP. A number was chosen which hopefully
makes port collisions highly unlikely.

Change-Id: Id7f82d8245401659d269e6ba084c2ad05aee068d
---
M tests/channels/pjsip/dialplan_functions/pjsip_channel/configs/ast1/extensions.conf
M tests/channels/pjsip/dialplan_functions/pjsip_channel/sipp/uac-no-hangup.xml
2 files changed, 3 insertions(+), 3 deletions(-)



  git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/60/8860/1

diff --git a/tests/channels/pjsip/dialplan_functions/pjsip_channel/configs/ast1/extensions.conf b/tests/channels/pjsip/dialplan_functions/pjsip_channel/configs/ast1/extensions.conf
index 22ec556..8d6b12d 100644
--- a/tests/channels/pjsip/dialplan_functions/pjsip_channel/configs/ast1/extensions.conf
+++ b/tests/channels/pjsip/dialplan_functions/pjsip_channel/configs/ast1/extensions.conf
@@ -22,13 +22,13 @@
 
 ; Source will often be various things; just make sure we get something back
 same => n,GoSub(default,test_variable,1(rtp,src,audio,!=,""))
-same => n,GoSub(default,test_variable,1(rtp,dest,audio,=,"127.0.0.1:6000"))
+same => n,GoSub(default,test_variable,1(rtp,dest,audio,=,"127.0.0.1:9050"))
 same => n,GoSub(default,test_variable,1(rtp,hold,audio,=,"0"))
 same => n,GoSub(default,test_variable,1(rtp,secure,audio,=,"0"))
 same => n,GoSub(default,test_variable,1(rtp,direct,audio,=,"(null)"))
 
 ; Verify audio is set by default
-same => n,GoSub(default,test_variable,1(rtp,dest,,=,"127.0.0.1:6000"))
+same => n,GoSub(default,test_variable,1(rtp,dest,,=,"127.0.0.1:9050"))
 
 ; No video stream, these should be empty
 same => n,GoSub(default,test_variable,1(rtp,src,video,=,""))
diff --git a/tests/channels/pjsip/dialplan_functions/pjsip_channel/sipp/uac-no-hangup.xml b/tests/channels/pjsip/dialplan_functions/pjsip_channel/sipp/uac-no-hangup.xml
index ef10798..b3c3c89 100644
--- a/tests/channels/pjsip/dialplan_functions/pjsip_channel/sipp/uac-no-hangup.xml
+++ b/tests/channels/pjsip/dialplan_functions/pjsip_channel/sipp/uac-no-hangup.xml
@@ -22,7 +22,7 @@
       s=-
       c=IN IP[media_ip_type] [media_ip]
       t=0 0
-      m=audio [media_port] RTP/AVP 0
+      m=audio 9050 RTP/AVP 0
       a=rtpmap:0 PCMU/8000
 
     ]]>

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Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-MessageType: newchange
Gerrit-Change-Id: Id7f82d8245401659d269e6ba084c2ad05aee068d
Gerrit-Change-Number: 8860
Gerrit-PatchSet: 1
Gerrit-Owner: Kevin Harwell <kharwell at digium.com>
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