Integration of Sipit24 branch, many tickets involved:

 - #793: AMR encoder should regard 'mode-set' param specified by remote decoder.
 - #831: Automatically switch to TCP transport when sending large request
 - #832: Support for outbound proxy setting without using Route header
 - #849: Modify conference audio switch behavior in connecting ports.
 - #850: Remove 'Require=replaces' param in 'Refer-To' header (in call transfer with replaces).
 - #851: Support for regular nomination in ICE
 - #852: --ip-addr support for IPv6 for media transport in pjsua
 - #854: Adding SOFTWARE attribute in all outgoing requests may cause compatibility problem with older STUN server (thanks Alexei Kuznetsov for the report)
 - #855: Bug in digit map frequencies for DTMF digits (thanks FCCH for the report)
 - #856: Put back the ICE candidate priority values according to the default values in the draft-mmusic-ice
 - #857: Support for ICE keep-alive with Binding indication
 - #858: Do not authenticate STUN 438 response
 - #859: AMR-WB format param in the SDP is not negotiated correctly.
 - #867: Return error instead of asserting when PJSUA-LIB fails to open log file




git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2724 74dad513-b988-da41-8d7b-12977e46ad98

1 files changed, 1 insertions, 1 deletions

diff --git a/pjsip-apps/src/samples/siprtp_report.c b/pjsip-apps/src/samples/siprtp_report.c
index e63ef949..84c00724 100644
--- a/pjsip-apps/src/samples/siprtp_report.c
+++ b/pjsip-apps/src/samples/siprtp_report.c
@@ -77,7 +77,7 @@ static void print_call(int call_index)
 
     /* Call identification */
     len = pjsip_hdr_print_on(dlg->remote.info, userinfo, sizeof(userinfo));
-    if (len < 1)
+    if (len < 0)
 	pj_ansi_strcpy(userinfo, "<--uri too long-->");
     else
 	userinfo[len] = '\0';