Telephony
(→Session Initiation Protocol(SIP)) |
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| − | + | |style="width:500px; vertical-align:top; border:1px solid #fb9508;border-right:none;background-color:#fef0de;"|'''About SIP''' | |
| − | | style="width:500px; vertical-align:top; border:1px solid #fb9508;border-right:none;background-color:#fef0de;"|'''About SIP''' | + | |
|style="background-color:#fef0de;width:500px;border:1px solid #fb9508;vertical-align:top;"|'''SIP Features''' | |style="background-color:#fef0de;width:500px;border:1px solid #fb9508;vertical-align:top;"|'''SIP Features''' | ||
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<font class="tel-desc">SIP protocol in Yate</font><br/> | <font class="tel-desc">SIP protocol in Yate</font><br/> | ||
<div class="bullet"> </div><font class="tel-title">'''[[SIP Security in Yate]]'''</font><br/> | <div class="bullet"> </div><font class="tel-title">'''[[SIP Security in Yate]]'''</font><br/> | ||
| − | + | <font class="tel-desc">TLS and SRTP in Yate</font><br/> | |
| − | ;[[SIP Configuration File]] | + | <div class="bullet"> </div><font class="tel-title">'''[[SIP Configuration File]]</font><br/> |
| − | + | <font class="tel-desc">Main configuration file for SIP module in Yate.</font><br/> | |
| − | < | + | <div class="bullet"> </div><font class="tel-title"><font class="tel-desc">'''[[SIP Methods]]'''</font><br/> |
| − | + | <font class="tel-desc">How Yate processes SIP request methods and how to enable methods that are not handled by default.</font><br/> | |
| − | + | <div class="bullet"> </div><font class="tel-title">'''[[SIP Features Module]]'''</font><br/> | |
| − | + | <font class="tel-desc">SIP features module that implements SUBSCRIBE and NOTIFY Methods.</font><br/> | |
| − | ;[[SIP Features Module]] | + | <div class="bullet"> </div><font class="tel-title">'''[[SIP Routing in Yate]]</font><br/> |
| − | + | <font class="tel-desc">Route to a SIP channel and a SIP Line.</font><br/> | |
| − | ;[[SIP Routing in Yate]] | + | |
| − | + | ||
;[[SIP Client]] | ;[[SIP Client]] | ||
Revision as of 13:40, 26 November 2012
Yate is a telephony engine who's main function is to connect together call legs to external devices or to internal server resources. It uses a variety of VoIP protocols(as SIP, H.323, IAX2 or Jingle) that can be used without the need of special hardware. In the next chapters you will find more informations about configuring each telephony protocol and the features supported by Yate.
Contents |
Session Initiation Protocol(SIP)
The Session Initiation Protocol(SIP) is an signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol(IP).
| About SIP | SIP Features |
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SIP protocol in Yate TLS and SRTP in Yate Main configuration file for SIP module in Yate. How Yate processes SIP request methods and how to enable methods that are not handled by default. SIP features module that implements SUBSCRIBE and NOTIFY Methods. Route to a SIP channel and a SIP Line.
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H323
H.323 is a standard that specifies the components, protocols and procedures that provide multimedia communication services - real-time audio, video, and data communications - over packet-based networks.
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Media Gateway Control Protocol(MGCP)
Media Gateway Control Protocol is a signalling and call control protocol used within Voice over IP (VoIP) systems that typically inter-operate with the public switched telephone network (PSTN).
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Inter-Asterisk eXchange(IAX/IAX2)
IAX2 is a VoIP protocol that carries both signaling and media on the same port.
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JINGLE
Jingle is an extension to the Extensible Messaging and Presence Protocol (XMPP) which adds peer-to-peer (P2P) session control (signaling) for multimedia interactions such as in Voice over IP (VoIP) or videoconferencing communications.
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JABBER or XMPP(Extensible Messaging and Presence Protocol)
Extensible Messaging and Presence Protocol(XMPP) is a communications protocol for message-oriented middleware based on XML(Extensible Markup Language).
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See also
