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). | ||
| − | {|class=" | + | {|class="yate-table" |
|class="telephony_header_left"|'''About''' | |class="telephony_header_left"|'''About''' | ||
|class="telephony_header_right"|'''Features''' | |class="telephony_header_right"|'''Features''' | ||
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|class="telephony-content-left"| | |class="telephony-content-left"| | ||
| − | + | * [[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 | |
| + | |||
|class="telephony-content-right"| | |class="telephony-content-right"| | ||
| − | + | * [[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 NAT|SIP with NAT]] | |
| − | + | Resolving SIP traversal problem by Yate. | |
| − | + | * [[How_to_setup_chat_and_short_file_transfer_using_MESSAGE_Request_Method|How to use MESSAGE and PUBLISH requests]] | |
| − | + | Example on how to handle custom SIP requests. | |
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Revision as of 13:04, 11 September 2013
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 | 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. Implementation and SIP Client features in Yate |
SIP features module that implements SUBSCRIBE and NOTIFY methods. How to do configurations related to DTMFs in SIP channel. How to configure Yate to handle a SIP attended call transfer for an unknown call leg in the cluster node. Yate provides a protection mechanism against several types of SIP flood attacks. Describes how Yate can be used as a SIP session border controller. Use this message to initiate the transmission of a SIP request. Feature used in a routing module that allows Yate to query CNAM and LNP databases over the SIP protocol. Resolving SIP traversal problem by Yate. Example on how to handle custom SIP requests. |
H323 Protocol
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 | Features |
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Configuration file used to configure H323 protocol in Yate
Installing and compiling OpenH323 - stable version to use with Yate
How to compile Yate with H323plus support - unstable version to use with Yate
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Configure H323 Gatekeeper And Multiple Endpoint Server
Configure in Yate H323 To SIP Signalling Proxy
Describes DTMFs send related to h323 channel.
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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).
| About | Features |
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This module implements the Call Agent part of MGCP allowing Yate to control analog or digital Media Gateways
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Inter-Asterisk eXchange(IAX/IAX2)
IAX2 is a VoIP protocol that carries both signaling and media on the same port.
| About | Features |
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Main configuration file for IAX channel module in Yate.
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Setup to configure Yate as a IAX Server and YateClient as a IAX client
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JINGLE Protocol
Jingle - Google talk/Google voice.
| About | Features |
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How to configure JINGLE module in Yate
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How to configure a Jabber Client with Jingle Yate Server
Learn how to route calls to contacts using Gmail account in accfile.conf
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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).
| About | Features |
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Configuring Yate as a Jabber Client
Configuring Yate as a Jabber Server
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The features for Jabber Server
Describes the steps to follow to build a Jabber server using Yate.
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Miscellaneous
Various articles that apply to more than one protocol
| Articles |
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Description of call end causes and errors in various protocols
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See also
