
webrtc.sipthor.net
SylkWebRTC client for SylkServer
http://webrtc.sipthor.net/
WebRTC client for SylkServer
http://webrtc.sipthor.net/
TODAY'S RATING
>1,000,000
Date Range
HIGHEST TRAFFIC ON
Monday
LOAD TIME
2.9 seconds
16x16
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0
SSL
EXTERNAL LINKS
7
SITE IP
81.23.228.137
LOAD TIME
2.922 sec
SCORE
6.2
Sylk | webrtc.sipthor.net Reviews
https://webrtc.sipthor.net
WebRTC client for SylkServer
saghul | Mi Brain-Training Personal
http://saghul.net/blog/author/admin
Para que no se me olviden las cosas…. Archivo del Autor: saghul. Building Telephony Systems with OpenSIPS 2a edición. Hoy mismo se acaba de publicar el libro. Building Telephony Systems with OpenSIPS. En su segunda edición. Podéis comprarlo tanto en formato digital como en formato árbol muerto en la página de PacktPub. El libro es una actualización de la primera edición, utilizando OpenSIPS 2.X. La nueva rama de desarrollo de OpenSIPS. Esta entrada se publicó en General. Y está etiquetada con OpenSIPS.
General | Mi Brain-Training Personal
http://saghul.net/blog/category/general
Para que no se me olviden las cosas…. Archivo de la categoría: General. Building Telephony Systems with OpenSIPS 2a edición. Hoy mismo se acaba de publicar el libro. Building Telephony Systems with OpenSIPS. En su segunda edición. Podéis comprarlo tanto en formato digital como en formato árbol muerto en la página de PacktPub. El libro es una actualización de la primera edición, utilizando OpenSIPS 2.X. La nueva rama de desarrollo de OpenSIPS. Esta entrada se publicó en General. Espero que os sirva! 2015 ...
VoIP | Mi Brain-Training Personal
http://saghul.net/blog/tag/voip
Para que no se me olviden las cosas…. Archivo de la etiqueta: VoIP. Building Telephony Systems with OpenSIPS 2a edición. Hoy mismo se acaba de publicar el libro. Building Telephony Systems with OpenSIPS. En su segunda edición. Podéis comprarlo tanto en formato digital como en formato árbol muerto en la página de PacktPub. El libro es una actualización de la primera edición, utilizando OpenSIPS 2.X. La nueva rama de desarrollo de OpenSIPS. Esta entrada se publicó en General. Y está etiquetada con OpenSIPS.
WikiStart - SIP2SIP - AG Projects
http://wiki.sip2sip.info/projects/sip2sip/wiki
SIP2SIP is a real time communications service for Audio, Video, Presence, Chat, File Transfer and multiparty conferencing based on SIP signaling and related media protocols (RTP, MSRP and XCAP). The service is open and federates with external SIP and XMPP domains. Fair use policy ¶. The service is provided with no guarantee based on a fair use policy. Fair use is regulated by an automatic system that blocks any account that exceeds ten times the usage of an average subscriber. SIP domain enrollment ¶.
News - AG Projects
http://projects.ag-projects.com/news
SylkServer 3.3.0 released. Added by Adrian Georgescu. Sylkserver (3.3.0) unstable; urgency=medium. Webrtcgateway: refactor API message handling. Webrtcgateway: add support for setting account display name. Webrtcgateway: remove obsolete sylkrtc test application. Webrtcgateway: add ability to customize User Agent. Raised Janus version dependency. SylkServer 3.2.0 released. Added by Adrian Georgescu. Sylkserver (3.2.0) unstable; urgency=medium. Fix per-room pstn access numbers setting. Added command line o...
SipTesting - SIP2SIP - AG Projects
http://wiki.sip2sip.info/projects/sip2sip/wiki/SipTesting
Using your SIP account. Calls from Web browsers. Calls from PSTN network. Calls to PSTN network. IM and File Transfer. Using your SIP account ¶. First configure your SIP device. Incoming calls ¶. When you enroll a SIP account on SIP2SIP infrastructure, a SIP address under @sip2sip.info domain is automatically allocated to you. You must provide this SIP address to those that want to reach you. Then you must have your SIP device registered with SIP2SIP infrastructure. Calls from Web browsers ¶. You can cal...
SIP | Mi Brain-Training Personal
http://saghul.net/blog/tag/sip
Para que no se me olviden las cosas…. Archivo de la etiqueta: SIP. Building Telephony Systems with OpenSIPS 2a edición. Hoy mismo se acaba de publicar el libro. Building Telephony Systems with OpenSIPS. En su segunda edición. Podéis comprarlo tanto en formato digital como en formato árbol muerto en la página de PacktPub. El libro es una actualización de la primera edición, utilizando OpenSIPS 2.X. La nueva rama de desarrollo de OpenSIPS. Esta entrada se publicó en General. Y está etiquetada con OpenSIPS.
TOTAL LINKS TO THIS WEBSITE
7
Home - Web Phone click to call based on WebRTC
Turns website visitors into customers. Convert your online leads into sales. Livehelp on your website. Route calls to your contact center. Receive calls on mobile. Land inbound calls to computer. NITS Technology in VoIP. Decrease your Monthly telecomunication bills up to 80%. 100% free calls between local and international offices. Allow to you connect resources in multiple locations. Welcome to Web RTC. An innovative Real Time Communication. Solution, enabling your online web visitors to call you.
WebRTC
WebRTC is a free, open project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose. To enable rich, high quality, RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols. On March 24, 2015,. WebRTC moved the master source repository over to Git at. Live demos can be accessed at.
Home - Web Phone click to call based on WebRTC
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Home - Web Phone click to call based on WebRTC
Click to Call Benefits! Let your Website Visitors Click an instant Free Calls. Ring to multiple lines at once from your Web Page button. Add a Live Agent to your Online E-commerce Website. Easily embedded link in your Web Pages and Social Media. Get calls 100% FREE. Click to Call Us FREE. Route inbound calls from anyone to anywhere. Receive calls from your website to your mobile. Route landline calls to any device. Get a local, Toll-Free or Vanity Phone Number. Click to Call Us FREE! WebRTC is like web p...
GISNet+
และประชานาม ยพ ท กษ. ผ บร หารสำน กงานบรรเทาท กข. และประชานาม ยพ ท กษ. ฝ ายบรรเทาท กข ผ ประสบภ ย. ศ นย ปฎ บ ต การภ ยพ บ ต สภากาชาดไทย. ศ นย จ กษ มาตรและแว นตา สภากาชาดไทย. สถาน กาชาดห วห น เฉล มพระเก ยรต. สถาน กาชาดส ร นธร. สถาน กาชาดส ร นธร พ งงา. ฝ ายบร หารงานท วไป. ฝ ายการเง นและบ ญช. ฝ ายพ สด ทางการแพทย และท วไป. หมวดโทรศ พท สำค ญ. หมายเลขโทรศ พท ฉ กเฉ น.
Sylk
Dialer container
Classifieds demo, click to call. 1 bedroom terraced house for sale. Emperor's Gate, South Kensington, London. This property is marketed by Solaiemes Ltd estate agency. Crafted Property Agents - Brisbane. Browns Plains, Qld 4118. Crafted Property Agents - Brisbane. Browns Plains, Qld 4118. 124 Abbey Street Forestdale Qld 4118. Subscribe to our newsletter to be the first to find out about our latest properties.
WebRTC-Demo
Um was geht es hier? Diese Seite demonstriert die Verwendung der neuen Internet-Technologie WebRTC (Web Real-Time Communications) für Video-Telefonie und Datenübertragung. Diese ermöglicht es Browsern, über eine Netzwerkverbindung direkt mit andern Browsern zu kommunizieren (Peer-to-Peer Verbindung). Wie stelle ich eine Verbindung her? Sobald eine weitere Person denselben Kanal wählt, können Sie sich per Audio/Video und Chat austauschen. HTW Chur, IKT.
Home - Webcall click to call based on WebRTC
Turns website visitors into customers. Convert your online leads into sales. Livehelp on your website. Route calls to your contact center. Receive calls on mobile. Land inbound calls to computer. NITS Technology in VoIP. Decrease your Monthly telecomunication bills up to 80%. 100% free calls between local and international offices. Allow to you connect resources in multiple locations. Welcome to Web RTC. An innovative Real Time Communication. Solution, enabling your online web visitors to call you.
Login | TSecureVideo.com
TSecureVideo.com - Infinity. Person or conference to dial.
WebRTC Ventures - Custom Application Design & Development
Call Us: 1.617.871.0882. Custom WebRTC Medical Applications. Custom Mobile WebRTC Apps. O’Reilly WebRTC Course. We are experts in WebRTC and other real time technologies. Serving our customers in industries such as medical, education, financial, collaboration, and more! We build online video and real time communications experiences for web and mobile,. We are experts in WebRTC and other real time technologies. We build online video and real time communications experiences for web and mobile,. Our team of...