한국어

소프트스위치

온누리070 플레이스토어 다운로드
    acrobits softphone
     온누리 070 카카오 프러스 친구추가온누리 070 카카오 프러스 친구추가친추
     카카오톡 채팅 상담 카카오톡 채팅 상담카톡
    
     라인상담
     라인으로 공유

     페북공유

   ◎위챗 : speedseoul


  
     PAYPAL
     
     PRICE
     

pixel.gif

    before pay call 0088 from app


Welcome to SIPp

SIPp is a free Open Source test tool / traffic generator for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC and UAS) and establishes and releases multiple calls with the INVITE and BYE methods. It can also reads custom XML scenario files describing from very simple to complex call flows. It features the dynamic display of statistics about running tests (call rate, round trip delay, and message statistics), periodic CSV statistics dumps, TCP and UDP over multiple sockets or multiplexed with retransmission management and dynamically adjustable call rates.

Other advanced features include support of IPv6, TLS, SIP authentication, conditional scenarios, UDP retransmissions, error robustness (call timeout, protocol defense), call specific variable, Posix regular expression to extract and re-inject any protocol fields, custom actions (log, system command exec, call stop) on message receive, field injection from external CSV file to emulate live users.

SIPp can also send media (RTP) traffic through RTP echo and RTP / pcap replay. Media can be audio or audio+video.

While optimized for traffic, stress and performance testing, SIPp can be used to run one single call and exit, providing a passed/failed verdict.

Last, but not least, SIPp has a comprehensive documentation available both in HTML and PDF format.

SIPp can be used to test many real SIP equipements like SIP proxies, B2BUAs, SIP media servers, SIP/x gateways, SIP PBX, ... It is also very useful to emulate thousands of user agents calling your SIP system.

Here is a screenshot:

SIPp screenshot

And here is a video of SIPp in action (Windows Media Player 9 codec or above required):

wmvsipp-01.wmv

Want to know more? Please jump to the documentation section.

조회 수 :
135573
등록일 :
2011.12.23
23:08:35 (*.160.42.233)
엮인글 :
http://webs.co.kr/index.php?document_srl=541&act=trackback&key=c11
게시글 주소 :
http://webs.co.kr/index.php?document_srl=541
List of Articles
번호 제목 글쓴이 날짜 조회 수
18 Asterisk Installation Asterisk Realtime configuration admin 2013-04-06 45287
17 The SIP Router Project admin 2013-04-06 44591
16 Kamailio :: A Quick Introduction admin 2013-04-06 80826
15 Welcome to the Smartvox Knowledgebase admin 2013-04-06 103874
14 Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration using Asterisk Database admin 2013-04-06 51907
13 OpenSIPS vs Asterisk admin 2013-04-06 218131
12 OpenSER_from_an_asterisk_POV file admin 2013-04-06 38365
11 Using SIP Devices behind NAT OPensip Asterisk IPPhone SIP Telephony file admin 2013-03-31 223689
10 rfc5766-turn-server admin 2013-03-21 40855
9 OpenSIPS Kick Start‎: VIDEO admin 2013-02-20 38247
8 OPENSIP Training VIDEO admin 2013-02-20 36596
7 What is new in 1.8.0 opensip admin 2012-05-21 250774
6 Asterisk v1.4x built on FreeBSD v7.1 UNIX admin 2012-01-06 148186
» SIP 트래픽 생성 테스트 툴 admin 2011-12-23 135573
4 사설 망 환경에서 SIP 의 NAT Traversal 문제 admin 2011-12-23 142793
3 the OpenSIPS Project OpenSIP admin 2011-12-14 44962
2 OpenH323 Gatekeeper - The GNU Gatekeeper admin 2011-12-14 53013
1 The FreeRADIUS Project admin 2011-12-14 43499