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


https://code.google.com/p/telepresence/


http://www.excitingip.com/4156/telepresence-open-source-sip-telepresencemcu/


http://conf-call.org/technical-guide.pdf?svn=2


http://www.medooze.com/products/mcu/open-source-installation.aspx


http://130.238.130.111/seminars/workshop-2011-03-31/minisip_mar31_workshop.pdf




Main features

This is a short but not exhaustive list of supported features on this beta version:

  • Powerful MCU (Multipoint Control Unit) for audio and video mixing
  • Stereoscopic (spatial) 3D and stereophonic audio
  • Full (1080p) and Ultra (2160p) HD video up to 120fps
  • Conference recording to a file (containers: .mp4.avi.mkv or .webm)
  • Revolutionary way to share presentations: documents are "streamed" in the video channel to allow any SIP client running on any device to participate
  • Smart adaptive audio and video bandwidth management
  • Congestion control mechanism
  • SIP registrar
  • 4 SIP transports (WebSocketTCPTLS and UDP)
  • SA (direct connection to SIP clients) and AS (behind a server, such as AsteriskreSIProcateopenSIPSKamailio…) modes
  • Support for any WebRTC-capable browser (WebRTC demo client at http://conf-call.org/)
  • Mixing different audio and video codecs on a single bridge (h264vp8, h263, mp4v-es, theora, opusg711, speex, g722, gsm, g729, amr, ilbc)
  • Protecting a bridge with PIN code
  • Unlimited number of bridges and participants
  • Connecting any SIP client (Mobiles, Tablets, Desktops, Set-top-boxes, Smart TVs...)
  • Easy interconnection with PSTN
  • NAT traversal (Symmetric RTP, RTCP-MUX, ICE, STUN and TURN)
  • RTCP Feedbacks (NACK, PLI, FIR, TMMBN, REMB…) for better video experience
  • Secure signalling (WSS, TLS) and media (SDES-SRTP and DTLS-SRTP)
  • Continuous presence
  • Smart algorithm to detect speakers and listeners
  • Different video patterns/layouts
  • Multiple operating systems (LinuxOS XWindows …)
  • 100% open source and free (no locked features)
  • Full documentation
  • …and many others

This short list is a good starting point to help you to understand what you could expect from our Telepresence system.

Getting started

  1. Read the technical guide for more information on how to buildinstall and run the system
  2. Test the system as explained here
  3. Share issues and technical questions on our developer group
  4. Find our roadmap here

Even if any SIP client could be used we highly recommend for this beta version to use our WebRTC demo client to ease debugging.

Technical help

Please check our issue tracker or developer group if you have any problem. 

We highly recommend reading our Technical guide

Please check the list of known issues before reporting.

조회 수 :
190248
등록일 :
2014.03.12
20:06:33 (*.251.139.148)
엮인글 :
http://webs.co.kr/index.php?document_srl=39244&act=trackback&key=ab0
게시글 주소 :
http://webs.co.kr/index.php?document_srl=39244
List of Articles
번호 제목 글쓴이 날짜 조회 수sort
172 OpenSIPS Control Panel (OCP) Installation Guide admin 2014-03-06 285227
171 Opensips Gateway between SIP and SMPP messages admin 2019-02-19 271939
170 What is new in 1.8.0 opensip admin 2012-05-21 260100
169 What is new in 2.3.0 opensips admin 2017-09-04 249567
168 Using SIP Devices behind NAT OPensip Asterisk IPPhone SIP Telephony file admin 2013-03-31 232445
167 OpenSIPS vs Asterisk admin 2013-04-06 227198
166 PUSH SERVER 푸시서버 안드로이드 애플 admin 2017-09-11 213795
» telepresence: Open Source SIP Telepresence/MCU admin 2014-03-12 190248
164 MediaProxy Installation Guide admin 2014-03-06 187387
163 RTPproxy Frequentry Asked Questions (FAQ) ¶ admin 2014-03-07 180547
162 SIP PBX - OpenSIPS and Asterisk configuration admin 2014-03-12 170682
161 Asterisk v1.4x built on FreeBSD v7.1 UNIX admin 2012-01-06 156610
160 사설 망 환경에서 SIP 의 NAT Traversal 문제 admin 2011-12-23 151269
159 How to install Mediaproxy 2.5.2 on CentOS 6 64 bit admin 2017-09-04 147551
158 SIP 트래픽 생성 테스트 툴 admin 2011-12-23 142084
157 opensips command /sbin/opensipsctl detail admin 2017-09-04 129218
156 How to setup a Jabber / XMPP server on Debian 8 (jessie) using ejabberd admin 2017-09-13 128260
155 Opensips_1.9 install guide this is great I like this admin 2014-03-04 112223
154 OpenSIPS basic configuration script 기본 컨피그 admin 2017-09-05 109474
153 Installing RTPproxy Start RTPproxy in Bridged mode very good admin 2014-03-07 108394
152 Welcome to the Smartvox Knowledgebase admin 2013-04-06 107346
151 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅 admin 2014-08-11 105543
150 Flooding Asterisk, Freeswitch and Kamailio with Metasploit admin 2013-04-06 104029
149 OpenSIPS Control Panel install guide admin 2014-03-06 101240
148 dictionary.opensips radius admin 2017-12-09 100669
147 Installation and configuration process record opensips 1.9.1 admin 2014-08-09 98368
146 in opensips what is lookup(domain [, flags [, aor]]) admin 2017-12-09 96630
145 My new toy: Bluebox-ng admin 2013-04-06 94033
144 Conference Support in Kamailio (OpenSER) admin 2014-03-12 91537
143 Kamailo OpenSIPs installation on Debian admin 2014-03-09 87944