sipcapture

https://github.com/sipcapture

https://github.com/sipcapture/homer/wiki/Examples%3A-FreeSwitch

 

FreeSwitch

FreeSWITCH Capture Agent

Freeswitch ships with an integrated HEP Capture Agent designed to work with HOMER

**FreeSwitch HEP3/EEP support is available in 1.6.8+ **

Global Configuration

To enable HEP capturing, open sofia.conf.xml and set capture-server param

<param name="capture-server" value="udp:192.168.0.1:9060"/>

Freeswitch 1.7 (master git) has support for HEPv2 and HEPv3. The new syntax is:

<param name="capture-server" value="udp:192.168.0.1:9060;hep=3;capture_id=100"/>

open internal.xml and change sip-capture param to "yes"

<param name="sip-capture" value="yes"/>

note: the ip address and port must be same as the listen param in your kamailio.cfg


To enable/disable the HEP agent on demand, you can use CLI commands:

freeswitch@fsnode04> sofia global capture on
 
+OK Global capture on
freeswitch@fsnode04> sofia global capture off
 
+OK Global capture off

Profile Configuration

You can choose to activate HEP capturing only for a specific profile:

freeswitch@fsnode04> sofia profile internal capture on
 
Enabled sip capturing on internal

freeswitch@fsnode04> sofia profile internal capture off
 
Disabled sip capturing on internal

B2BUA Correlation

To correlate B2BUA legs set the following before bridging the second leg:

      <action application="set" data="sip_h_X-cid=${sip_call_id}"/>

ESL Integration (beta)

hepipe.js provides experimental support for FreeSWITCH ESL integration for call quality reports feeding to HOMER 5, effectively providing external HEP3/EEP features with correlation support.

Events

ESL EventHep ModeHEP Type
CHANNEL_CREATELOG100
CHANNEL_ANSWERLOG100
CHANNEL_DESTROYLOG100
CUSTOMLOG100
RECV_RTCP_MESSAGERTCP5
CHANNEL_DESTROYCUSTOM QoS99

For full instructions and details please checkout hepipe.js

If you test or extend this feature please share your feedback!

Master the art of advanced VoIP and WebRTC communication with the most dynamic application server, FreeSWITCH About This Book Forget the hassle - make FreeSWITCH work for you Discover how FreeSWITCH integrates with a range of tools and APIs From high availability to IVR development use this book to become more confident with this useful communication software Who This Book Is For SysAdmins, VoIP engineers – whoever you are, whatever you’re trying to do, this book will help you get more from FreeSWITCH. What You Will Learn Get to grips with the core concepts of FreeSWITCH Learn FreeSWITCH high availability Work with SIP profiles, gateways, ITSPs, and Codecs optimization Implement effective security on your projects Master audio manipulation and recording Discover how FreeSWITCH works alongside WebRTC Build your own complex IVR and PBX applications Connect directly to PSTN/TDM Create your own FreeSWITCH module Trace SIP packets with the help of best open source tools Implement Homer Sipcapture to troubleshoot and debug all your platform traffic In Detail FreeSWITCH is one of the best tools around if you’re looking for a modern method of managing communication protocols through a range of different media. From real-time browser communication with the WebRTC API to implementing VoIP (voice over internet protocol), with FreeSWITCH you’re in full control of your projects. This book shows you how to unlock its full potential – more than just a tutorial, it’s packed with plenty of tips and tricks to make it work for you. Written by members of the team who actually helped build FreeSWITCH, it will guide you through some of the newest features of version 1.6 including video transcoding and conferencing. Find out how FreeSWITCH interacts with other tools and APIs, learn how to tackle common (and not so common) challenges ranging from high availability to IVR development and programming advanced PBXs. Great communication functionality begins with FreeSWITCH – find out how and get your project up and running today. Style and approach Find out how it works, then put your knowledge into practice - that's how this advanced FreeSWITCH guide has been designed to help you learn. You'll soon master FreeSWITCH and be confident using it in your projects. Table of Contents Chapter 1: Typical Voice Uses for FreeSWITCH Chapter 2: Deploying FreeSWITCH Chapter 3: ITSP and Voice Codecs Optimization Chapter 4: VoIP Security Chapter 5: Audio File and Streaming Formats, Music on Hold, Recording Calls Chapter 6: PSTN and TDM Chapter 7: WebRTC and Mod_Verto Chapter 8: Audio and Video Conferencing Chapter 9: Faxing and T38 Chapter 10: Advanced IVR with Lua Chapter 11: Write Your FreeSWITCH Module in C Chapter 12: Tracing and Debugging VoIP Chapter 13: Homer, Monitoring and Troubleshooting your Communication Platform
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