Telephony

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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).
 
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).
  
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Revision as of 13:24, 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
 
;SIP in Yate
SIP in Yate
SIP protocol in Yate
SIP Security in Yate
TLS and SRTP in Yate
SIP Configuration File
Main configuration file for SIP module in Yate.
SIP Methods
How Yate processes SIP request methods and how to enable methods that are not handled by default.
SIP Features Module
SIP features module that implements SUBSCRIBE and NOTIFY Methods.
SIP Routing in Yate
Route to a SIP channel and a SIP Line.
SIP Client
Implementation and SIP Client features in Yate
SIP Features
SIP Features Module
SIP features module that implements SUBSCRIBE and NOTIFY methods
SIP Send DTMFs
How to do configurations related to DTMFs in SIP channel.
SIP Attended Call Transfer In Cluster
How to configure Yate to handle a SIP attended call transfer for an unknown call leg in the cluster node
SIP Flood Protection
Yate provides a protection mechanism against several types of SIP flood attacks.
SIP SBC
Describes how Yate can be used as a SIP session border controller.
xsip.generate
Use this message to initiate the transmission of a SIP request.
SIP query for CNAM and LNP
Feature used in a routing module that allows Yate to query CNAM and LNP databases over the SIP protocol.
SIP with NAT
Resolving SIP traversal problem by Yate.



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.

About H323

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).

About MGCP

Inter-Asterisk eXchange(IAX/IAX2)

IAX2 is a VoIP protocol that carries both signaling and media on the same port.

About IAX

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.

About JINGLE

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).

About Jabber

See also

Retrieved from "https://docs.yate.ro/wiki/index.php?title=Telephony&oldid=2385"
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