Alternatives and similar repositories for sipcmd

Users that are interested in sipcmd are comparing it to the libraries listed below

• sipcapture / sipgrep
  SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console
  ☆169Updated 2 months ago
• voxserv / freeswitch_conf_minimal
  Minimalistic FreeSWITCH configuration as a start for new projects
  ☆82Updated 8 years ago
• voxserv / freeswitch-perf-dialer
  FreeSWITCH dialer program for VoIP performance tests
  ☆46Updated 7 years ago
• zaf / asterisk-googletts
  Asterisk AGI script that uses Google's translate text to speech service.
  ☆219Updated last year
• zaf / asterisk-speech-recog
  Speech recognition script for Asterisk that uses google's speech engine.
  ☆252Updated 7 years ago
• mojolingo / sippy_cup
  Create SIP load test scenarios the easy way
  ☆223Updated 4 years ago
• sippy / b2bua
  Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.
  ☆188Updated last month
• nils-ohlmeier / sipsak
  SIP swiss army knife
  ☆162Updated 2 months ago
• andrius / asterisk
  Asterisk PBX in Docker on Alpine Linux. Smallest Asterisk ever! ;)
  ☆262Updated 2 years ago
• vir / yate
  Yet Another Telephony Engine - UNOFFICIAL mirror
  ☆114Updated 7 years ago
• provisioner / Provisioner
  The is the central location for the Provisioner Module for VoIP/PBX Servers. Most of the new work is happening inside the v5-dev branch
  ☆123Updated last year
• jchavanton / voip_patrol
  VoIP signaling and media test automation
  ☆119Updated last month
• asipto / siremis-php
  Web Admin Interface for Kamailio
  ☆103Updated 5 months ago
• miconda / sipexer
  Modern and flexible SIP/VoIP cli tool
  ☆354Updated last month
• havfo / WEBRTC-to-SIP
  Setup for a WEBRTC client and Kamailio server to call SIP clients
  ☆297Updated last year
• ghrst / Kamailio-HA
  This repository contains Ansible playbooks and related files for an Active-Passive Kamailio auto-deployment using Pacemaker and Corosync.
  ☆55Updated 6 years ago
• meetecho / asterisk-opus
  Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration
  ☆81Updated 10 years ago
• dOpensource / dsiprouter
  UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services
  ☆211Updated last week
• AGProjects / python-sipsimple
  SIP SIMPLE implementation for Python
  ☆333Updated 4 years ago
• etamme / federated-sip
  Federated SIP deployment
  ☆35Updated last year
• saghul / sipp-scenarios
  SIPp scenarios I use for testing SIP stuff
  ☆293Updated last year
• SIPp / pysipp
  SIPp for Humans - launch multiple agents with Python
  ☆187Updated 9 months ago
• irontec / sngrep
  Ncurses SIP Messages flow viewer
  ☆1,104Updated last month
• voipmonitor / sniffer
  VoIPmonitor sniffer sources
  ☆255Updated this week
• pbertera / SIPp-by-example
  SIPp examples
  ☆137Updated last year
• sems-server / sems
  Sip Express Media Server
  ☆172Updated 3 months ago
• sippy / rtpproxy
  The RTPproxy is a high-performance software proxy for RTP streams that can work together with Sippy B2BUA, Kamailio, OpenSIPS and SER.
  ☆447Updated 3 weeks ago
• sipcapture / captagent
  100% Open-Source Packet Capture Agent for HEP
  ☆173Updated 3 months ago
• staskobzar / amiws
  Asterisk Management Interface (AMI) to Web-socket proxy
  ☆87Updated 2 years ago
• drachtio / drachtio-server
  A SIP call processing server that can be controlled via nodejs applications
  ☆286Updated 3 weeks ago