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From: Kaushik K. A. <mai...@gm...> - 2019-05-07 17:32:20
|
Need to test the following scenario in IMS env with 1000(2534-3533) users: Configuration Scenario: sipp1<-->TAS<--->sipp2(different machine) Call Flow scenario: Register + Invite with media from sipp1 to sipp2 via TAS. My queries: 1- In this scenario do we need to run both the sipp instance in uac mode OR 1st one as uac and 2nd one as UAS. 2-How I can send simultaneous 1000 registrations+invite from different numbers? 3-Please provide me a sample script and csv for this scenario (if possible) with commands to run. Thank you. |
|
From: Jeannot L. <Jea...@mi...> - 2018-05-10 19:52:14
|
FYI: After much troubleshooting I have managed to resolve this problem on my own: it is my environment -- not the build -- that had an issue. I had installed the following tools after building them from sources (since the versions provided by default on my CentOS 6.9 box were no longer supported by SIPP): autoconf 2.69 automake 1.13.4 (These two above were installed in the typical custom /usr/local path) I turned out to have the CentOS 6.9-provided "pkgconfig" package installed - but in the default /usr/bin location: pkg-config 0.23 As it turns out autoconf was unable to find pkg-config in a "/usr/local"-based location that it expected. ==> So the solution for me was to proceed with the installation of pkg-config 0.29.2 (latest version) from sources and into the custom /usr/local path; this allowed autoconf to find pkg-config happily. -- Jeannot Langlois Software Developer MiVoice Border Gateway Development Mitel Networks 350 Legget Drive, Kanata, Ontario K2K 2W7 http://www.mitel.com<http://www.mitel.com/> Jea...@mi...<mailto:Jea...@mi...> (613) 592-5660 x74420 (613) 691-3385 [Direct Dial] " It's not over until I win. " -- Leslie Brown From: Jeannot Langlois [mailto:Jea...@mi...] Sent: Wednesday, May 09, 2018 5:26 PM To: sip...@li... Subject: [Sipp-devel] Broken build: "configure.ac:99: error: possibly undefined macro: AC_SEARCH_LIBS" / "configure.ac:100: error: possibly undefined macro: AC_ERROR" Hi there, I'm trying to build the latest SIPP 3.6 dev code (April 30th 2018) from the master branch of the GIT repository "https://github.com/SIPp/sipp.git". It appears as if the build got recently broken: " $ ./build.sh --with-rtpstream --with-openssl Submodule 'gmock' (https://chromium.googlesource.com/external/googlemock) registered for path 'gmock' Submodule 'gtest' (https://chromium.googlesource.com/external/googletest) registered for path 'gtest' Initialized empty Git repository in /home/fsuser/sipp-git/gmock/.git/ remote: Total 2287 (delta 1682), reused 2287 (delta 1682) Receiving objects: 100% (2287/2287), 700.99 KiB, done. Resolving deltas: 100% (1682/1682), done. Submodule path 'gmock': checked out 'c7ee6b5c206e498063b1e89d381045f78cb6ab36' Initialized empty Git repository in /home/fsuser/sipp-git/gtest/.git/ remote: Total 4343 (delta 3384), reused 4343 (delta 3384) Receiving objects: 100% (4343/4343), 1.19 MiB, done. Resolving deltas: 100% (3384/3384), done. Submodule path 'gtest': checked out '935f1265d088d81e802c49aec8cb717c30741caa' autoreconf: Entering directory `.' autoreconf: configure.ac: not using Gettext autoreconf: running: aclocal --force autoreconf: configure.ac: tracing autoreconf: configure.ac: not using Libtool autoreconf: running: /usr/local/bin/autoconf --force configure.ac:99: error: possibly undefined macro: AC_SEARCH_LIBS If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. configure.ac:100: error: possibly undefined macro: AC_ERROR autoreconf: /usr/local/bin/autoconf failed with exit status: 1 " I am using the following: CentOS: v6.9. gcc/g++: v4.4.7 pkg-config: v0.23 autoconf: v2.69 automake: v1.13.4 I am suspecting this to have been introduced by a recent patch from April 23rd: " commit 90f38e465062b6a57326269085f4511b3dae2c3d Author: Walter Doekes <wal...@wj...> Date: Mon Apr 23 10:29:54 2018 +0200 Use pkg-config to get ncurses/tinfo libs if available Fixes #271 reported by @a-tinsmith. diff --git a/configure.ac b/configure.ac index f1fd79a..92d04ef 100644 --- a/configure.ac +++ b/configure.ac @@ -84,13 +84,18 @@ case "$host" in ;; esac +AC_CHECK_PROG([have_pkgconfig], [pkg-config], [yes], [no]) + # ==================== checks for libraries ============================= -AC_CHECK_LIB(curses,initscr) -if test x$ac_cv_lib_curses_initscr = xno; then - AC_CHECK_LIB(ncurses,initscr) - if test x$ac_cv_lib_ncurses_initscr = xno; then - AC_MSG_ERROR([ncurses library missing]) - fi + +if test "$have_pkgconfig" = "yes"; then + # Use pkg-config when available + PKG_PROG_PKG_CONFIG() + PKG_CHECK_MODULES([ncurses], [ncurses], LIBS="$LIBS $ncurses_LIBS") +else + # Olden ways + AC_SEARCH_LIBS([initscr], [ncurses curses],,AC_ERROR([Missing (n)curses library])) + AC_SEARCH_LIBS([stdscr], [tinfo ncurses curses],,AC_ERROR([Missing (n)curses/tinfo library])) fi AC_CHECK_LIB(pthread, pthread_mutex_init, THREAD_LIBS="-lpthread", @@ -182,7 +187,6 @@ AM_CONDITIONAL(HAVE_RTP, test "$rtp" = "yes") # Conditional build with gsl if test "$gsl" = "yes"; then - AC_CHECK_PROG([have_pkgconfig], [pkg-config], [yes], [no]) if test "$have_pkgconfig" != "yes" ; then AC_MSG_ERROR([Please install pkg-config to use the gsl libraries]) fi diff --git a/include/config.h.in b/include/config.h.in index 38ee7fc..b883dca 100644 --- a/include/config.h.in +++ b/include/config.h.in @@ -70,18 +70,12 @@ /* Define to 1 if you have the `crypto' library (-lcrypto). */ #undef HAVE_LIBCRYPTO -/* Define to 1 if you have the `curses' library (-lcurses). */ -#undef HAVE_LIBCURSES - /* Define to 1 if you have the `gslcblas' library (-lgslcblas). */ #undef HAVE_LIBGSLCBLAS /* Define to 1 if you have the `m' library (-lm). */ #undef HAVE_LIBM -/* Define to 1 if you have the `ncurses' library (-lncurses). */ -#undef HAVE_LIBNCURSES - /* Define to 1 if you have the `ssl' library (-lssl). */ #undef HAVE_LIBSSL " ==> Unfortunately I am not very knowledgeable about autoconf/automake/pkg-config to figure out the problem by myself alone - at least not without any clues. ==> Can anyone give a clue as to why this build break occurs? Even better - can anyone submit a fix? Let me know if I need to provide more information - or if you want me to try anything. Thanks! :) -- Jeannot Langlois Software Developer MiVoice Border Gateway Development Mitel Networks 350 Legget Drive, Kanata, Ontario K2K 2W7 http://www.mitel.com<http://www.mitel.com/> Jea...@mi...<mailto:Jea...@mi...> (613) 592-5660 x74420 (613) 691-3385 [Direct Dial] " It's not over until I win. " -- Leslie Brown ________________________________ NOTE: This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message. |
|
From: Jeannot L. <Jea...@mi...> - 2018-05-09 22:59:52
|
Hi there, I'm trying to build the latest SIPP 3.6 dev code (April 30th 2018) from the master branch of the GIT repository "https://github.com/SIPp/sipp.git". It appears as if the build got recently broken: " $ ./build.sh --with-rtpstream --with-openssl Submodule 'gmock' (https://chromium.googlesource.com/external/googlemock) registered for path 'gmock' Submodule 'gtest' (https://chromium.googlesource.com/external/googletest) registered for path 'gtest' Initialized empty Git repository in /home/fsuser/sipp-git/gmock/.git/ remote: Total 2287 (delta 1682), reused 2287 (delta 1682) Receiving objects: 100% (2287/2287), 700.99 KiB, done. Resolving deltas: 100% (1682/1682), done. Submodule path 'gmock': checked out 'c7ee6b5c206e498063b1e89d381045f78cb6ab36' Initialized empty Git repository in /home/fsuser/sipp-git/gtest/.git/ remote: Total 4343 (delta 3384), reused 4343 (delta 3384) Receiving objects: 100% (4343/4343), 1.19 MiB, done. Resolving deltas: 100% (3384/3384), done. Submodule path 'gtest': checked out '935f1265d088d81e802c49aec8cb717c30741caa' autoreconf: Entering directory `.' autoreconf: configure.ac: not using Gettext autoreconf: running: aclocal --force autoreconf: configure.ac: tracing autoreconf: configure.ac: not using Libtool autoreconf: running: /usr/local/bin/autoconf --force configure.ac:99: error: possibly undefined macro: AC_SEARCH_LIBS If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. configure.ac:100: error: possibly undefined macro: AC_ERROR autoreconf: /usr/local/bin/autoconf failed with exit status: 1 " I am using the following: CentOS: v6.9. gcc/g++: v4.4.7 pkg-config: v0.23 autoconf: v2.69 automake: v1.13.4 I am suspecting this to have been introduced by a recent patch from April 23rd: " commit 90f38e465062b6a57326269085f4511b3dae2c3d Author: Walter Doekes <wal...@wj...> Date: Mon Apr 23 10:29:54 2018 +0200 Use pkg-config to get ncurses/tinfo libs if available Fixes #271 reported by @a-tinsmith. diff --git a/configure.ac b/configure.ac index f1fd79a..92d04ef 100644 --- a/configure.ac +++ b/configure.ac @@ -84,13 +84,18 @@ case "$host" in ;; esac +AC_CHECK_PROG([have_pkgconfig], [pkg-config], [yes], [no]) + # ==================== checks for libraries ============================= -AC_CHECK_LIB(curses,initscr) -if test x$ac_cv_lib_curses_initscr = xno; then - AC_CHECK_LIB(ncurses,initscr) - if test x$ac_cv_lib_ncurses_initscr = xno; then - AC_MSG_ERROR([ncurses library missing]) - fi + +if test "$have_pkgconfig" = "yes"; then + # Use pkg-config when available + PKG_PROG_PKG_CONFIG() + PKG_CHECK_MODULES([ncurses], [ncurses], LIBS="$LIBS $ncurses_LIBS") +else + # Olden ways + AC_SEARCH_LIBS([initscr], [ncurses curses],,AC_ERROR([Missing (n)curses library])) + AC_SEARCH_LIBS([stdscr], [tinfo ncurses curses],,AC_ERROR([Missing (n)curses/tinfo library])) fi AC_CHECK_LIB(pthread, pthread_mutex_init, THREAD_LIBS="-lpthread", @@ -182,7 +187,6 @@ AM_CONDITIONAL(HAVE_RTP, test "$rtp" = "yes") # Conditional build with gsl if test "$gsl" = "yes"; then - AC_CHECK_PROG([have_pkgconfig], [pkg-config], [yes], [no]) if test "$have_pkgconfig" != "yes" ; then AC_MSG_ERROR([Please install pkg-config to use the gsl libraries]) fi diff --git a/include/config.h.in b/include/config.h.in index 38ee7fc..b883dca 100644 --- a/include/config.h.in +++ b/include/config.h.in @@ -70,18 +70,12 @@ /* Define to 1 if you have the `crypto' library (-lcrypto). */ #undef HAVE_LIBCRYPTO -/* Define to 1 if you have the `curses' library (-lcurses). */ -#undef HAVE_LIBCURSES - /* Define to 1 if you have the `gslcblas' library (-lgslcblas). */ #undef HAVE_LIBGSLCBLAS /* Define to 1 if you have the `m' library (-lm). */ #undef HAVE_LIBM -/* Define to 1 if you have the `ncurses' library (-lncurses). */ -#undef HAVE_LIBNCURSES - /* Define to 1 if you have the `ssl' library (-lssl). */ #undef HAVE_LIBSSL " ==> Unfortunately I am not very knowledgeable about autoconf/automake/pkg-config to figure out the problem by myself alone - at least not without any clues. ==> Can anyone give a clue as to why this build break occurs? Even better - can anyone submit a fix? Let me know if I need to provide more information - or if you want me to try anything. Thanks! :) -- Jeannot Langlois Software Developer MiVoice Border Gateway Development Mitel Networks 350 Legget Drive, Kanata, Ontario K2K 2W7 http://www.mitel.com<http://www.mitel.com/> Jea...@mi...<mailto:Jea...@mi...> (613) 592-5660 x74420 (613) 691-3385 [Direct Dial] " It's not over until I win. " -- Leslie Brown ________________________________ NOTE: This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message. |
|
From: Sunny M. <apo...@ho...> - 2017-03-01 02:02:48
|
Hi Dev group, I wonder whether Sipp will support OPUS later? Or how possibly to make it support OPUS? Thanks, Sunny Mok Sent from Outlook<http://aka.ms/weboutlook> |
|
From: Sakharam T. <sak...@ou...> - 2016-10-19 08:41:14
|
Hi All, I have extract 00FA08271007F558071CCB value from below header using regex, Refer-To: <sip:42...@hr...?User-to-User=00FA08271007F558071CCB%3bencoding%3dhex> What regex i should use ? Best Regards, Sakharam Thorat. |
|
From: <nir...@wi...> - 2016-02-03 10:02:26
|
Hi, Actually play_dtmf support. In the sipp-v3.4.1 version, play_dtmf exec support is not there. I need sipp release which includes the play_dtmf support patch. Thanks & Regards, Nirmala V. ________________________________ From: Sakharam Thorat <sak...@ou...> Sent: Wednesday, February 3, 2016 2:53 PM To: Nirmala V (NEP); sip...@li... Subject: Re: [Sipp-devel] Regarding Run-time RFC2833 DTMF generation patch - need of official sipp release https://github.com/SIPp/sipp/archive/v3.4.1.tar.gz Best Regards, Sakharam Thorat. ________________________________ From: nir...@wi... <nir...@wi...> Sent: Wednesday, February 3, 2016 1:57 PM To: sip...@li... Subject: [Sipp-devel] Regarding Run-time RFC2833 DTMF generation patch - need of official sipp release Hi Team, I am using sipp application (sipp.svn) version SIPp v3.2-PCAP, version unknown, built Dec 3 2011, 11:34:51.). For testing the dtmf, I have applied the patch attached in the link http://sourceforge.net/p/sipp/patches/50/, [http://a.fsdn.com/allura/nf/1453400641/_ew_/theme/sftheme/images/sftheme/logo-black-svg_g.png]<http://sourceforge.net/p/sipp/patches/50/> sipp / Patches / #50 Run-time RFC2833 DTMF generation patch<http://sourceforge.net/p/sipp/patches/50/> sourceforge.net The patch allows to send inband DTMF codes to remote party over RTP using RTP EVENT messages described in RFC2833 by specifying the following expressions in the XML ... When I use xml script with play_dtmf exec, sipp application got segmentation fault. Then I have made the below changes in prepare_pcap.c file. 1. In free_pkts(pcap_pkts *pkts) function pkt_index is initialized (pcap_pkt *pkt_index = pkts->pkts and it is incrementated (pkt_index++) inside while loop. 2. In the rtphdr structure both timestamp and ssrcid changed to u_int32_t. After that sipp runs successfully with play_dtmf. Can you provide the official sipp application release which includes the dtmf patch and which also includes the aboves changes. Thanks & Regards, Nirmala V. |
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From: Sakharam T. <sak...@ou...> - 2016-02-03 09:23:39
|
https://github.com/SIPp/sipp/archive/v3.4.1.tar.gz Best Regards, Sakharam Thorat. ________________________________ From: nir...@wi... <nir...@wi...> Sent: Wednesday, February 3, 2016 1:57 PM To: sip...@li... Subject: [Sipp-devel] Regarding Run-time RFC2833 DTMF generation patch - need of official sipp release Hi Team, I am using sipp application (sipp.svn) version SIPp v3.2-PCAP, version unknown, built Dec 3 2011, 11:34:51.). For testing the dtmf, I have applied the patch attached in the link http://sourceforge.net/p/sipp/patches/50/, [http://a.fsdn.com/allura/nf/1453400641/_ew_/theme/sftheme/images/sftheme/logo-black-svg_g.png]<http://sourceforge.net/p/sipp/patches/50/> sipp / Patches / #50 Run-time RFC2833 DTMF generation patch<http://sourceforge.net/p/sipp/patches/50/> sourceforge.net The patch allows to send inband DTMF codes to remote party over RTP using RTP EVENT messages described in RFC2833 by specifying the following expressions in the XML ... When I use xml script with play_dtmf exec, sipp application got segmentation fault. Then I have made the below changes in prepare_pcap.c file. 1. In free_pkts(pcap_pkts *pkts) function pkt_index is initialized (pcap_pkt *pkt_index = pkts->pkts and it is incrementated (pkt_index++) inside while loop. 2. In the rtphdr structure both timestamp and ssrcid changed to u_int32_t. After that sipp runs successfully with play_dtmf. Can you provide the official sipp application release which includes the dtmf patch and which also includes the aboves changes. Thanks & Regards, Nirmala V. |
|
From: <nir...@wi...> - 2016-02-03 08:43:36
|
Hi Team, I am using sipp application (sipp.svn) version SIPp v3.2-PCAP, version unknown, built Dec 3 2011, 11:34:51.). For testing the dtmf, I have applied the patch attached in the link http://sourceforge.net/p/sipp/patches/50/, [http://a.fsdn.com/allura/nf/1453400641/_ew_/theme/sftheme/images/sftheme/logo-black-svg_g.png]<http://sourceforge.net/p/sipp/patches/50/> sipp / Patches / #50 Run-time RFC2833 DTMF generation patch<http://sourceforge.net/p/sipp/patches/50/> sourceforge.net The patch allows to send inband DTMF codes to remote party over RTP using RTP EVENT messages described in RFC2833 by specifying the following expressions in the XML ... When I use xml script with play_dtmf exec, sipp application got segmentation fault. Then I have made the below changes in prepare_pcap.c file. 1. In free_pkts(pcap_pkts *pkts) function pkt_index is initialized (pcap_pkt *pkt_index = pkts->pkts and it is incrementated (pkt_index++) inside while loop. 2. In the rtphdr structure both timestamp and ssrcid changed to u_int32_t. After that sipp runs successfully with play_dtmf. Can you provide the official sipp application release which includes the dtmf patch and which also includes the aboves changes. Thanks & Regards, Nirmala V. |
|
From: Guillaume C. <gco...@gm...> - 2015-11-09 20:23:09
|
diff -Naur sipp-3.4.1_old/include/logger.hpp sipp-3.4.1_new/include/logger.hpp
--- sipp-3.4.1_old/include/logger.hpp 2014-03-09 15:04:57.000000000 -0400
+++ sipp-3.4.1_new/include/logger.hpp 2015-10-29 14:42:21.980132542 -0400
@@ -24,6 +24,7 @@
extern bool useShortMessagef _DEFVAL(0);
extern bool useScreenf _DEFVAL(0);
extern bool useLogf _DEFVAL(0);
+extern bool useLogMessagef _DEFVAL(0);
//extern bool useTimeoutf _DEFVAL(0);
extern bool dumpInFile _DEFVAL(0);
extern bool dumpInRtt _DEFVAL(0);
diff -Naur sipp-3.4.1_old/src/sipp.cpp sipp-3.4.1_new/src/sipp.cpp
--- sipp-3.4.1_old/src/sipp.cpp 2014-03-09 15:04:57.000000000 -0400
+++ sipp-3.4.1_new/src/sipp.cpp 2015-10-29 14:42:21.990137542 -0400
@@ -363,6 +363,7 @@
{"trace_logs", "Allow tracing of <log> actions in <scenario file name>_<pid>_logs.log.", SIPP_OPTION_SETFLAG, &useLogf, 1},
+ {"trace_msg_logs", "Displays sent and received SIP messages in <scenario file name>_<pid>_logs.log. Require trace_logs to be enabled.", SIPP_OPTION_SETFLAG, &useLogMessagef, 1},
{"log_file", "Set the name of the log actions log file.", SIPP_OPTION_LFNAME, &log_lfi, 1},
{"log_overwrite", "Overwrite the log actions log file (default true).", SIPP_OPTION_LFOVERWRITE, &log_lfi, 1},
diff -Naur sipp-3.4.1_old/src/socket.cpp sipp-3.4.1_new/src/socket.cpp
--- sipp-3.4.1_old/src/socket.cpp 2014-03-09 15:04:57.000000000 -0400
+++ sipp-3.4.1_new/src/socket.cpp 2015-10-29 14:42:21.993139043 -0400
@@ -1100,6 +1100,12 @@
struct timeval currentTime;
GET_TIME (¤tTime);
+ if (useLogMessagef == 1)
+ {
+ LOG_MSG("%s:%s <-- Received %s (CSeq: %s)\n",
+ CStat::formatTime(¤tTime), call_id, get_first_line(msg), get_header_content(msg, "CSeq:"));
+ }
+
if (useShortMessagef == 1) {
TRACE_SHORTMSG("%s\tR\t%s\tCSeq:%s\t%s\n",
CStat::formatTime(¤tTime),call_id, get_header_content(msg,"CSeq:"), get_first_line(msg));
@@ -2370,6 +2376,14 @@
CStat::formatTime(¤tTime), call_id, get_header_content(msg,"CSeq:"), get_first_line(msg));
free(msg);
}
+
+ if (useLogMessagef == 1)
+ {
+ char *msg = strdup(buffer);
+ char *call_id = get_call_id(msg);
+ LOG_MSG("%s:%s --> Sending %s (CSeq: %s)\n", CStat::formatTime(¤tTime), call_id, get_first_line(msg), get_header_content(msg, "CSeq:"));
+ free(msg);
+ }
} else if (rc <= 0) {
if ((errno == EWOULDBLOCK) && (flags & WS_BUFFER)) { |
|
From: Vijay G. <vg...@ct...> - 2015-11-09 18:45:59
|
Hi, Just want to know whether SIPp supports SRV record? Say if I place a call to _sip._udp.lab.local which is pointing to a DNS name which is resolving to an ipaddress. E.g sipp _sip._udp.lab.local -p 5062 -t u1 -inf test.csv -r 1 -sf scenario.xml -min_rtp_port 15000 -max_rtp_port 25000 -m 1 -nd -skip_rlimit I am getting "Unknown remote host '_sip._udp.lab.local' ". Did SRV records worked for anyone? Or is it supported in another version or patch? Regards, Vijay. |
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From: Mititelu S. <fa...@gm...> - 2015-05-02 16:35:18
|
Hello, I am new to SIPp code. I am thinking of the following .xml scenario: send 1000 INVITEs pause 999999999999 send 1000 reINVITEs What I want to do is unpause the message flow using a new command "echo cset pause 0 > /dev/udp/0.0.0.0/8.8.8.9"; this is supposed to overwrite the current huge pause. This is helpful when one wants to control the message flow from outside. I think to make scenario::runInit() return a call* to have a call reference and refer it from process_set() (in socket.cpp file). What would be the best thing to do? Do you have any suggestions? Regards, Stefan |
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From: Yuriy G. <ovo...@gm...> - 2014-12-20 09:16:32
|
Hello. I already wrote this letter to sipp-users list but it not answered
at 3 days. I use latest sipp and standart test scenario for test RTP at my
system
./sipp -s 1007 mysystem.ip:5060 -sf scenarios/sipp_uac_pcap_g711a.xml -m 1
-mi 10.0.1.41 -mp 30010 -d 1200 -l 1 -r 1 -rp 10 -trace_rtt -trace_err
-stat_delimiter ,
at sipp_uac_pcap_g711a.xml I use standart string to start g711a.pcap file
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="g711a.pcap"/>
</action>
</nop>
<!-- Pause 90 seconds, which is approximately the duration of the -->
<!-- PCAP file -->
<pause milliseconds="90000"/>
everething works fine but audio plays only after 20 seconds delay. I check
RTP at my asterisk. IT start on 2 legs after 200 OK message. So how I cat
remove 20 seconds delay?
Also I tried to play rtp_stream:
I set convered 8Hz mono file with 16000 bitrate format at my scenario to
200OK recv fileld:
<recv response="200" rtd="true" crlf="true">
<action>
<exec rtp_stream="hello-world.wav,1,8"/>
</action>
When I start my scenario- i see error
Cannot read/cache rtpstream file
|
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From: Tais P. H. <ta...@os...> - 2014-09-23 08:51:50
|
Hi, Daily builds of SIPp for Ubuntu are available via Launchpad the link below: https://launchpad.net/~taisph/+archive/ubuntu/sipp -- Tais Plougmann Hansen |
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From: Sakharam T. <sak...@ou...> - 2014-09-23 07:30:49
|
I compiled Detailed procedure about installation at, http://techvick.blogspot.in/2014/09/how-to-intsall-sipp-on-ubuntu.html if you face any issue later can take a look at there. Best Regards,Sakharam Thorat. > From: kum...@NE... > To: kum...@gm... > Date: Fri, 19 Sep 2014 15:29:14 +0000 > CC: sip...@li... > Subject: Re: [Sipp-users] compile sipp with Openssl > > How did you complied SIPp? > > > On 19 Sep 2014, at 17:54, kumar uppu <kum...@gm...> wrote: > > > HI all > > > > I am unable to compile sipp with openssl > > > > i did below process > > 1)installed sipp-3.3 and untar that one > > 2)followed all the instruction through the sipp document > > > > but i got below error > > > > 1411138112.307593: To use a TLS transport you must compile SIPp with OpenSSL > > > > > > > > Can any one help please > > > > > > With Regards > > Kumar > > ------------------------------------------------------------------------------ > > Slashdot TV. Video for Nerds. Stuff that Matters. > > http://pubads.g.doubleclick.net/gampad/clk?id=160591471&iu=/4140/ostg.clktrk_______________________________________________ > > Sipp-users mailing list > > Sip...@li... > > https://lists.sourceforge.net/lists/listinfo/sipp-users > > Bu e-posta mesajı ve ekleri gönderildiği kişi ya da kuruma özeldir ve gizlidir. Ayrıca hukuken de gizli olabilir. Hiçbir şekilde üçüncü kişilere açıklanamaz ve yayınlanamaz. Eğer mesajın gönderildiği alıcı değilseniz bu elektronik postanın içeriğini açıklamanız, kopyalamanız, yönlendirmeniz ve kullanmanız kesinlikle yasaktır ve bu elektronik postayı ve eklerini derhal silmeniz gerekmektedir. NETAŞ TELEKOMÜNİKASYON A.Ş. bu mesajın içerdiği bilgilerin doğruluğu veya eksiksiz olduğu konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin ne şekilde olursa olsun içeriğinden, iletilmesinden, alınmasından, saklanmasından ve kullanılmasından sorumlu değildir. Bu mesajdaki görüşler gönderen kişiye ait olup, NETAŞ TELEKOMÜNİKASYON A.Ş.’nin görüşlerini yansıtmayabilir. > ------------------------------------------------------- > This e-mail and its attachments are private and confidential and intended for the exclusive use of the individual or entity to whom it is addressed. It may also be legally confidential. Any disclosure, distribution or other dissemination of this message to any third party is strictly prohibited. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited, and the e-mail should immediately be deleted. NETAŞ TELEKOMÜNİKASYON A.Ş. makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the transmission, reception, storage or use of such information in any way whatsoever. The opinions expressed in this message are those of the sender and may not necessarily reflect the opinions of NETAŞ TELEKOMÜNİKASYON A.Ş. > ------------------------------------------------------------------------------ > Slashdot TV. Video for Nerds. Stuff that Matters. > http://pubads.g.doubleclick.net/gampad/clk?id=160591471&iu=/4140/ostg.clktrk > _______________________________________________ > Sipp-users mailing list > Sip...@li... > https://lists.sourceforge.net/lists/listinfo/sipp-users |
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From: Vijay G. <vg...@ct...> - 2014-07-29 15:08:35
|
When I ran the sipp(client) on the same server where sipserver is running I see sip and rtp traffic passing through local interface(lo: 127.0.0.1) even though I used the ipaddress(10.10.x.x) of the machine passing as (-i and -mi) using pcap play but when I ran sipp on a different server than the sipserver I see only sip traffic on the local ipaddress(10.10.x.x) of the sipserver but don't see the rtp traffic. I ran this problem long back when I was trying to use sipp 3.3 then I moved to rtpstream instead of pcapplay as it was just audio at that time. Now as the rtp stream don't support playing video I will be using pcapplay to play audio and video but seeing the same issue again with 3.4.1. Can anyone please let me know how I can send media on a different ipaddress? Here is the command I am using sipp 10.10.20.15 -p 5062 -t u1 -inf callervijay.csv -r 1 -sf test.xml -i 10.10.20.17 -mi 10.10.20.17 -mp 6000 -min_rtp_port 30000 -max_rtp_port 60000 -m 1 -nd -skip_rlimit Regards, Vijay. |
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From: Vijay G. <vg...@ct...> - 2014-07-07 15:11:50
|
Hi ,
Does rtp stream support playing video?
I know we can play rtp audio stream as
<action>
<exec rtp_stream="pcap/rtp.ulaw,1,0"/>
</action>
How can I play video stream?
If rtpstream don't support playing video can anyone let me know how I can play video stream using pcap play?
Regards,
Vijay.
|
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From: Gaurav <gau...@gm...> - 2014-06-10 09:59:38
|
Hi, I'd like to file a bug reported by the clang static analyzer. Description: Address of stack memory associated with local variable 'infos' returned to caller File: src/socket.cpp Line: 2878 Variable T_peer_infos infos; is declared on stack, and it will be automatically released when current block goes out of the scope. I think we can allocate infos in the caller function and pass it to get_peer_socket() function. -gaurav |
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From: Gaurav <gau...@gm...> - 2014-06-10 09:13:15
|
Hi,
I'd like to file a bug reported by the clang static analyzer.
Description: Dereference of null pointer (loaded from variable 'port')
File: src/socket.cpp
Line: 1462
After moving the following code (pointer check) block at 1481 to the
beginning of the function sipp_bind_socket() fixes this bug (when USE_SCTP
is defined).
if (!port) {
return 0;
}
-Gaurav Nangla
|
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From: Vijay G. <vg...@ct...> - 2014-05-13 20:13:47
|
Does sip support playing amr,amr-wb rtp stream? I know it does support g711 ulaw and alaw. If rtpstream does not, does pcap_play support amr pcap play? If yes how can I achieve that? Regards, Vijay. |
|
From: Vijay G. <vg...@ct...> - 2014-03-28 14:50:47
|
That would be great. Thanks Rob. Regards, Vijay. -----Original Message----- From: rob...@gm... [mailto:rob...@gm...] On Behalf Of Rob Day Sent: Friday, March 28, 2014 10:47 AM To: Vijay Goje Cc: sipp-users; sipp-devel Subject: Re: [Sipp-users] jSIPp - a ground-up rewrite of SIPp to make feature development easier Hi Vijay, I haven't implemented any media handling yet, but it's definitely high on my list - I know SIPp is pretty weak on the media side (especially in terms of analysing the media and providing stats) and I'm hoping to improve on that now I have this smaller, more flexible codebase. I'd expect to implement some RTP/RTCP support in the next few weeks. Best, Rob On 28 March 2014 14:01, Vijay Goje <vg...@ct...> wrote: > Hi Rob, > > I am looking forward for this. > Does the jSIPp support rtpstream play? > > Regards, > Vijay. > > -----Original Message----- > From: Rob Day [mailto:rk...@rk...] > Sent: Thursday, March 27, 2014 7:31 PM > To: sipp-users; sipp-devel > Subject: [Sipp-users] jSIPp - a ground-up rewrite of SIPp to make > feature development easier > > Hi all, > > I've been experimenting recently with a ground-up rewrite of SIPp, to simplify the code, make it more portable, easier to integrate with other tools, and faster to develop new features for. I expect to still maintain the existing C++ version, but the complexity of the code makes real innovation (as opposed to bugfixes and small improvements) difficult. > > This has resulted in what I'm currently calling jSIPp, implemented in about 1500 lines of Java compared to the 22,000 lines of C++ SIPp currently uses. The page for it is https://github.com/rkday/jsipp, with a download at https://github.com/rkday/jsipp/releases/tag/v0.0.4. > As a Java JAR file, it should (in theory) run on any operating system with Java installed. > > It's still alpha quality (current version 0.0.4), but it can already process simple call scenarios at high rates. It uses the same or similar XML files and command-line arguments as SIPp, but doesn't support all the features yet - https://github.com/rkday/jsipp/wiki/Current-Status documents exactly what is available. > > The most noticeable difference from SIPp is the UI - rather than having a built-in ncurses UI, jSIPp publishes the call events it sees over a ZeroMQ messaging socket, making it easy to build custom UIs and tools that record and display the performance data as required. I've created two UIs for it already, a traditional-looking ncurses one in Ruby and a more modern web UI with graphs. You can find the details at https://github.com/rkday/jsipp/wiki/ZeroMQ#sample-programs, along with enough information to build your own interfaces. > > If you'd like to see what new features I hope to add now that I have a simpler, cleaner implementation of SIPp, see https://github.com/rkday/jsipp/blob/master/future-directions.md. If you want to add features or fixes yourself, see https://github.com/rkday/jsipp/blob/master/design.md for a design overview to get you started. I'm happy to accept pull requests (https://help.github.com/articles/using-pull-requests). > > It would be really nice to get some early users using this for real projects - so if there are particular features you need implemented before you can use it, please let me know and I'll do what I can to prioritise them. Likewise, if you'd like to contribute but have questions or problems, let me know. > > jSIPp, like SIPp, is licensed under the GPL and free to use. > Feedback/comments/criticism are all welcome. > > Best, > Rob > > P.S. In case anyone is wondering what happened to the last new project I announced on this list - Quaff is still under active development (https://github.com/rkday/quaff), and likely to feature in my KamailioWorld talk about SIP testing next month. > > ---------------------------------------------------------------------- > -------- _______________________________________________ > Sipp-users mailing list > Sip...@li... > https://lists.sourceforge.net/lists/listinfo/sipp-users |
|
From: Rob D. <rk...@rk...> - 2014-03-28 14:47:19
|
Hi Vijay, I haven't implemented any media handling yet, but it's definitely high on my list - I know SIPp is pretty weak on the media side (especially in terms of analysing the media and providing stats) and I'm hoping to improve on that now I have this smaller, more flexible codebase. I'd expect to implement some RTP/RTCP support in the next few weeks. Best, Rob On 28 March 2014 14:01, Vijay Goje <vg...@ct...> wrote: > Hi Rob, > > I am looking forward for this. > Does the jSIPp support rtpstream play? > > Regards, > Vijay. > > -----Original Message----- > From: Rob Day [mailto:rk...@rk...] > Sent: Thursday, March 27, 2014 7:31 PM > To: sipp-users; sipp-devel > Subject: [Sipp-users] jSIPp - a ground-up rewrite of SIPp to make feature development easier > > Hi all, > > I've been experimenting recently with a ground-up rewrite of SIPp, to simplify the code, make it more portable, easier to integrate with other tools, and faster to develop new features for. I expect to still maintain the existing C++ version, but the complexity of the code makes real innovation (as opposed to bugfixes and small improvements) difficult. > > This has resulted in what I'm currently calling jSIPp, implemented in about 1500 lines of Java compared to the 22,000 lines of C++ SIPp currently uses. The page for it is https://github.com/rkday/jsipp, with a download at https://github.com/rkday/jsipp/releases/tag/v0.0.4. > As a Java JAR file, it should (in theory) run on any operating system with Java installed. > > It's still alpha quality (current version 0.0.4), but it can already process simple call scenarios at high rates. It uses the same or similar XML files and command-line arguments as SIPp, but doesn't support all the features yet - https://github.com/rkday/jsipp/wiki/Current-Status documents exactly what is available. > > The most noticeable difference from SIPp is the UI - rather than having a built-in ncurses UI, jSIPp publishes the call events it sees over a ZeroMQ messaging socket, making it easy to build custom UIs and tools that record and display the performance data as required. I've created two UIs for it already, a traditional-looking ncurses one in Ruby and a more modern web UI with graphs. You can find the details at https://github.com/rkday/jsipp/wiki/ZeroMQ#sample-programs, along with enough information to build your own interfaces. > > If you'd like to see what new features I hope to add now that I have a simpler, cleaner implementation of SIPp, see https://github.com/rkday/jsipp/blob/master/future-directions.md. If you want to add features or fixes yourself, see https://github.com/rkday/jsipp/blob/master/design.md for a design overview to get you started. I'm happy to accept pull requests (https://help.github.com/articles/using-pull-requests). > > It would be really nice to get some early users using this for real projects - so if there are particular features you need implemented before you can use it, please let me know and I'll do what I can to prioritise them. Likewise, if you'd like to contribute but have questions or problems, let me know. > > jSIPp, like SIPp, is licensed under the GPL and free to use. > Feedback/comments/criticism are all welcome. > > Best, > Rob > > P.S. In case anyone is wondering what happened to the last new project I announced on this list - Quaff is still under active development (https://github.com/rkday/quaff), and likely to feature in my KamailioWorld talk about SIP testing next month. > > ------------------------------------------------------------------------------ > _______________________________________________ > Sipp-users mailing list > Sip...@li... > https://lists.sourceforge.net/lists/listinfo/sipp-users |
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From: Vijay G. <vg...@ct...> - 2014-03-28 14:16:07
|
Hi Rob, I am looking forward for this. Does the jSIPp support rtpstream play? Regards, Vijay. -----Original Message----- From: Rob Day [mailto:rk...@rk...] Sent: Thursday, March 27, 2014 7:31 PM To: sipp-users; sipp-devel Subject: [Sipp-users] jSIPp - a ground-up rewrite of SIPp to make feature development easier Hi all, I've been experimenting recently with a ground-up rewrite of SIPp, to simplify the code, make it more portable, easier to integrate with other tools, and faster to develop new features for. I expect to still maintain the existing C++ version, but the complexity of the code makes real innovation (as opposed to bugfixes and small improvements) difficult. This has resulted in what I'm currently calling jSIPp, implemented in about 1500 lines of Java compared to the 22,000 lines of C++ SIPp currently uses. The page for it is https://github.com/rkday/jsipp, with a download at https://github.com/rkday/jsipp/releases/tag/v0.0.4. As a Java JAR file, it should (in theory) run on any operating system with Java installed. It's still alpha quality (current version 0.0.4), but it can already process simple call scenarios at high rates. It uses the same or similar XML files and command-line arguments as SIPp, but doesn't support all the features yet - https://github.com/rkday/jsipp/wiki/Current-Status documents exactly what is available. The most noticeable difference from SIPp is the UI - rather than having a built-in ncurses UI, jSIPp publishes the call events it sees over a ZeroMQ messaging socket, making it easy to build custom UIs and tools that record and display the performance data as required. I've created two UIs for it already, a traditional-looking ncurses one in Ruby and a more modern web UI with graphs. You can find the details at https://github.com/rkday/jsipp/wiki/ZeroMQ#sample-programs, along with enough information to build your own interfaces. If you'd like to see what new features I hope to add now that I have a simpler, cleaner implementation of SIPp, see https://github.com/rkday/jsipp/blob/master/future-directions.md. If you want to add features or fixes yourself, see https://github.com/rkday/jsipp/blob/master/design.md for a design overview to get you started. I'm happy to accept pull requests (https://help.github.com/articles/using-pull-requests). It would be really nice to get some early users using this for real projects - so if there are particular features you need implemented before you can use it, please let me know and I'll do what I can to prioritise them. Likewise, if you'd like to contribute but have questions or problems, let me know. jSIPp, like SIPp, is licensed under the GPL and free to use. Feedback/comments/criticism are all welcome. Best, Rob P.S. In case anyone is wondering what happened to the last new project I announced on this list - Quaff is still under active development (https://github.com/rkday/quaff), and likely to feature in my KamailioWorld talk about SIP testing next month. ------------------------------------------------------------------------------ _______________________________________________ Sipp-users mailing list Sip...@li... https://lists.sourceforge.net/lists/listinfo/sipp-users |
|
From: Rob D. <rk...@rk...> - 2014-03-27 23:31:00
|
Hi all, I've been experimenting recently with a ground-up rewrite of SIPp, to simplify the code, make it more portable, easier to integrate with other tools, and faster to develop new features for. I expect to still maintain the existing C++ version, but the complexity of the code makes real innovation (as opposed to bugfixes and small improvements) difficult. This has resulted in what I'm currently calling jSIPp, implemented in about 1500 lines of Java compared to the 22,000 lines of C++ SIPp currently uses. The page for it is https://github.com/rkday/jsipp, with a download at https://github.com/rkday/jsipp/releases/tag/v0.0.4. As a Java JAR file, it should (in theory) run on any operating system with Java installed. It's still alpha quality (current version 0.0.4), but it can already process simple call scenarios at high rates. It uses the same or similar XML files and command-line arguments as SIPp, but doesn't support all the features yet - https://github.com/rkday/jsipp/wiki/Current-Status documents exactly what is available. The most noticeable difference from SIPp is the UI - rather than having a built-in ncurses UI, jSIPp publishes the call events it sees over a ZeroMQ messaging socket, making it easy to build custom UIs and tools that record and display the performance data as required. I've created two UIs for it already, a traditional-looking ncurses one in Ruby and a more modern web UI with graphs. You can find the details at https://github.com/rkday/jsipp/wiki/ZeroMQ#sample-programs, along with enough information to build your own interfaces. If you'd like to see what new features I hope to add now that I have a simpler, cleaner implementation of SIPp, see https://github.com/rkday/jsipp/blob/master/future-directions.md. If you want to add features or fixes yourself, see https://github.com/rkday/jsipp/blob/master/design.md for a design overview to get you started. I'm happy to accept pull requests (https://help.github.com/articles/using-pull-requests). It would be really nice to get some early users using this for real projects - so if there are particular features you need implemented before you can use it, please let me know and I'll do what I can to prioritise them. Likewise, if you'd like to contribute but have questions or problems, let me know. jSIPp, like SIPp, is licensed under the GPL and free to use. Feedback/comments/criticism are all welcome. Best, Rob P.S. In case anyone is wondering what happened to the last new project I announced on this list - Quaff is still under active development (https://github.com/rkday/quaff), and likely to feature in my KamailioWorld talk about SIP testing next month. |
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From: Rob D. <rk...@rk...> - 2013-12-27 22:40:11
|
Thanks for this, Peter - I don't want to put anything new into v3.4 at this stage, but I'm taking a look at this as part of early v3.5 work. In particular, the media port stuff looks sensible and will probably go in; I'm going to experiment a bit with out-of-scenario messages and see if I can reproduce that memory leak, as I'd expect successful calls to call terminate(CStat::E_CALL_SUCCESSFULLY_ENDED) which calls 'delete this', so it surprises me a little. If you have existing scripts which you think might have exhibited the memory leak, it would be useful to see them. Thanks again! Rob On 19/12/13 13:05, samwise wrote: > Hi, > > I work on a team which has made a collection of changes to a local > branch of SIPp v3.3 to work with our local environment, after we > experienced some instability after running a series of very long tests > with parallel instances of SIPp (i.e. over several days). Some of > those team members have since left and my own C++ is ropey enough that > I'm not actually sure of all the ramifications of these changes. > > Anyway, I thought I'd submit the patches I have to the SIPp users > mailing list so that you can take a look through and see whether any > of them would benefit v3.4, as it appears to be on the way, rather > than us maintaining a private branch. Apologies if these duplicate > any other patch submissions / changes made since v3.3. > > I realise that some of these changes may not be appropriate for the > mainline release, but I'd be interested to know if you do decide to > take any of these patches to apply to the main release. > > Anyway, hope they're of some use. Thanks! > > Peter. > > 1. Change to Call-Id format of number-pid-random@ip to include random > number which will prevent Call-Id from being repeated > 2. Fix SIPp memory leak > 3. Fixing bug in SIPp that did not check if the video port it was > binding to was in use > 4. Increasing SIPp media port range > 5. Fix another SIPp memory leak > 6. Fix bug in Makefile when compiling with SCTP support > > > ------------------------------------------------------------------------------ > Rapidly troubleshoot problems before they affect your business. Most IT > organizations don't have a clear picture of how application performance > affects their revenue. With AppDynamics, you get 100% visibility into your > Java,.NET, & PHP application. Start your 15-day FREE TRIAL of AppDynamics Pro! > http://pubads.g.doubleclick.net/gampad/clk?id=84349831&iu=/4140/ostg.clktrk > > > _______________________________________________ > Sipp-devel mailing list > Sip...@li... > https://lists.sourceforge.net/lists/listinfo/sipp-devel |
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From: samwise <sam...@ba...> - 2013-12-19 13:37:42
|
Hi, I work on a team which has made a collection of changes to a local branch of SIPp v3.3 to work with our local environment, after we experienced some instability after running a series of very long tests with parallel instances of SIPp (i.e. over several days). Some of those team members have since left and my own C++ is ropey enough that I'm not actually sure of all the ramifications of these changes. Anyway, I thought I'd submit the patches I have to the SIPp users mailing list so that you can take a look through and see whether any of them would benefit v3.4, as it appears to be on the way, rather than us maintaining a private branch. Apologies if these duplicate any other patch submissions / changes made since v3.3. I realise that some of these changes may not be appropriate for the mainline release, but I'd be interested to know if you do decide to take any of these patches to apply to the main release. Anyway, hope they're of some use. Thanks! Peter. 1. Change to Call-Id format of number-pid-random@ip to include random number which will prevent Call-Id from being repeated 2. Fix SIPp memory leak 3. Fixing bug in SIPp that did not check if the video port it was binding to was in use 4. Increasing SIPp media port range 5. Fix another SIPp memory leak 6. Fix bug in Makefile when compiling with SCTP support |
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