* #27232: jni: added pjproject checkout as regular git content
We will remove it once the next release of pjsip (with Android support)
comes out and is merged into SFLphone.
diff --git a/jni/pjproject-android/.svn/pristine/3f/3f3a02bd48f9c6e9c966b4e158341f0d9d39fdf6.svn-base b/jni/pjproject-android/.svn/pristine/3f/3f3a02bd48f9c6e9c966b4e158341f0d9d39fdf6.svn-base
new file mode 100644
index 0000000..d34e4a9
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/3f/3f3a02bd48f9c6e9c966b4e158341f0d9d39fdf6.svn-base
@@ -0,0 +1,38 @@
+# $Id$
+import inc_sip as sip
+import inc_sdp as sdp
+
+# Ticket http://trac.pjsip.org/repos/ticket/718
+# RTC doesn't put rport in Via, and it is report to have caused segfault.
+complete_msg = \
+"""INVITE sip:localhost SIP/2.0
+Via: SIP/2.0/UDP $LOCAL_IP:$LOCAL_PORT;branch=z9hG4bK74a60ee5
+From: <sip:tester@localhost>;tag=as2858a32c
+To: <sip:pjsua@localhost>
+Contact: <sip:tester@$LOCAL_IP:$LOCAL_PORT>
+Call-ID: 123@localhost
+CSeq: 1 INVITE
+Max-Forwards: 70
+Content-Type: application/sdp
+Content-Length: 285
+
+v=0
+o=root 4236 4236 IN IP4 192.168.1.11
+s=session
+c=IN IP4 192.168.1.11
+t=0 0
+m=audio 14390 RTP/AVP 0 3 8 101
+a=rtpmap:0 PCMU/8000
+a=rtpmap:3 GSM/8000
+a=rtpmap:8 PCMA/8000
+a=rtpmap:101 telephone-event/8000
+a=fmtp:101 0-16
+a=silenceSupp:off - - - -
+a=ptime:20
+a=sendrecv
+"""
+
+
+sendto_cfg = sip.SendtoCfg( "RTC no rport", "--null-audio --auto-answer 200",
+ "", 200, complete_msg=complete_msg)
+