Skip to main content

OpenSource Telephony

Search This Blog

Twilio SIP Domains, FreeSWITCH integration

on September 08, 2024

Get link
Facebook
Twitter
Pinterest
Email
Other Apps

AI FreeSwitch SIP SIP API Telephony Twilio

Get link
Facebook
Twitter
Pinterest
Email
Other Apps

Comments

Post a Comment

Popular Posts

Step-by-Step Guide: Installing FreeSWITCH on Debian 12 "Bookworm".

FreeSWITCH is an open-source, scalable, and versatile telephony platform designed for building real-time communication applications and services. It provides a software-based alternative to traditional hardware-based PBX (Private Branch Exchange) systems and offers a wide range of communication features. In this comprehensive guide, we'll walk you through the process of installing and configuring FreeSWITCH on a Debian 12 server. Debian is known for its stability and reliability, making it an excellent choice for hosting your VoIP infrastructure. By the end of this tutorial, you'll have a fully functional FreeSWITCH installation ready to handle voice calls, video calls, and more. Step 1: Get Your SignalWire TOKEN Visit the SignalWire Website: Open your web browser and go to the SignalWire website (https://www.signalwire.com/). Sign Up or Log In: If you already have a SignalWire account, log in using your credentials. If not, sign up for a new account by following the registra

How to install SIPp ( SIP Performance testing tool) on Debian 11?

Install Pre-requisites sudo apt install build-essential libncurses-dev libssl-dev libpcap-dev libnet1 lksctp-tools libsctp-dev Download sipp source code and extract it. cd /usr/local/src/ wget -c https://github.com/SIPp/sipp/releases/download/v3.6.1/sipp-3.6.1.tar.gz tar -xvf sipp-3.6.1.tar.gz cd sipp-3.6.1/ Compile and make with TLS, PCAP Play and SCTP support. cmake . -DUSE_SSL=1 -DUSE_PCAP=1 -DUSE_SCTP=1 make

Transcoding using RTPEngine + Kamailio

RTPengine can be configured to perform transcoding, which involves converting digital audio and video files from one format to another in real-time as they are being transmitted over the network. This can be useful in situations where the devices on either end of the connection do not support the same codecs or where the network conditions require the use of a different codec to ensure the best possible quality of service. Kamailio configuration #!define WITH_NAT #!define WITH_RTPENGINE loadmodule "rtpengine.so" modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") route[RELAY] { if (has_body("application/sdp")) { if ($rU=="user1") { rtpengine_manage("codec-transcode-PCMU"); } else { rtpengine_manage("codec-transcode-opus"); } ... } onreply_route[MANAGE_REPLY] { if (has_body("application/sdp")) { rtpe

Powered by Blogger