Chat room for Linux / GNU / OpenSource talks using standard VOIP / SIP protocol

openbsd98324 · 06-22-2022, 01:01 PM

Hello,

here 10 clients for opensource talks...
https://www.slant.co/topics/6937/~vo...ents-for-linux

However, is there any single chat room ?

best regards.
James

More VOIP / SIP:

Code:

 apt-cache search voip | grep -v umble

Code:

captagent - HOMER SIP capture agent
coturn - TURN and STUN server for VoIP
ekiga - H.323 and SIP compatible VoIP client
ekiga-dbg - H.323 and SIP compatible VoIP client - debug symbols
ekiga-plugin-evolution - H.323 and SIP compatible VoIP client - evolution plugin
gsutil - configure and manage Grandstream BudgeTone 100 VOIP and GX2000 phones
homer-api - HOMER Capture Node REST API
homer-api-mysql - HOMER Capture Node REST API
homer-api-postgresql - HOMER Capture Node REST API
iprelay - User-space bandwidth shaping TCP proxy daemon
jitsi - VoIP and Instant Messaging client
libexosip2-11 - eXtended osip library
libexosip2-dev - eXtended osip library development files
libh323-1.24.0v5 - H.323 aka VoIP library
libh323-dbg - H.323 aka VoIP library development debug files
libh323plus-dev - H.323 aka VoIP library development files
libjs-sdp-transform - JavaScript parser/writer for Session Description Protocol
libnetsds-perl - Service Delivery Suite framework
libopal-dbg - OPAL library debug symbols
libopal-dev - OPAL library header files
libopal-doc - OPAL library documentation files
libopal3.10.10 - Open Phone Abstraction Library - successor of OpenH323
libpjmedia2 - PJ Project - VoIP media
libpjsua2 - PJ Project - Basic VoIP client library
libpjsua2-2v5 - PJ Project - Basic VoIP client library
libresiprocate-turn-client-1.11 - reSIProcate TURN client (reTurn) - shared libraries
libsipwitch-dev - secure peer-to-peer SIP VoIP server - development files
libsipwitch1 - secure peer-to-peer SIP VoIP server - shared libraries
libsipwitch1-dbg - secure peer-to-peer SIP VoIP server - debug symbols
libsofia-sip-ua-dev - Sofia-SIP library development files
libsofia-sip-ua-glib-dev - Sofia-SIP library glib/gobject interface development files
libsofia-sip-ua-glib3 - Sofia-SIP library glib/gobject interfaces runtime
libsofia-sip-ua0 - Sofia-SIP library runtime
libspeex-dev - The Speex codec library development files
libspeex-ocaml - OCaml interface to the speex library
libspeex-ocaml-dev - OCaml interface to the speex library
libspeex1 - The Speex codec runtime library
libspeexdsp-dev - The Speex extended library development files
libspeexdsp1 - The Speex extended runtime library
mangler - Ventrilo compatible client for Linux
mz - versatile packet creation and network traffic generation tool
node-rtcninja - JavaScript parser/writer for Session Description Protocol
node-sdp-transform - JavaScript parser/writer for Session Description Protocol
openam - H.323 answering machine
resiprocate-turn-server - reSIProcate SIP stack - ICE/TURN server
resiprocate-turn-server-psql - reSIProcate SIP stack - ICE/TURN server
sflphone-data - SIP and IAX2 compatible VoIP phone - common data
simpleopal - Simple example from the OPAL project
sip-tester - Performance testing tool for the SIP protocol
sipwitch - secure peer-to-peer VoIP server for the SIP protocol
sipwitch-cgi - secure peer-to-peer SIP VoIP server - CGI XML-RPC interface
sofia-sip-bin - Sofia-SIP library utilities
sofia-sip-doc - Sofia-SIP library documentation
speex - The Speex codec command line tools
speex-doc - Documentation for speex
turnserver - server for ICE/STUN/TURN, NAT traversal for SIP and Jabber
twinkle - Voice over Internet Protocol (VoIP) SIP Phone (GUI)
twinkle-common - Voice over Internet Protocol (VoIP) SIP Phone (common files)
twinkle-console - Voice over Internet Protocol (VoIP) SIP Phone (console)
asterisk-prompt-es - Spanish prompts for the Asterisk PBX

openbsd98324 · 06-25-2022, 06:48 AM

Do you maybe know a Echo test room to test my SIP Linphone ??

Turbocapitalist · 06-25-2022, 06:51 AM

There are 3333@sip2sip.info and 4444@sip2sip.info, if I recall correctly.

Depending on the amount of work you want to put in, you can also build an Asterisk server with the features you want.

openbsd98324 · 06-25-2022, 06:53 AM

Does this still work today??

sip:500@ekiga.net

Quote:

Ekiga’s numbers

SIPNumber Description

sip:500@ekiga.net Echo test, supports video (H264, H263-1998, H261 only), as well as audio (PCMA only)
sip:520@ekiga.net Call-me test (call, hangup, and you will be called right afterwards), supports audio only
sip:501@ekiga.net Public conference room from Ekiga.net
sip:501xxxx@ekiga.net These rooms are public or private, anyone can join a conference at any time if he choose the right number or you can protect the access with a PIN number.
sip:5011122@ekiga.net french conference room 1122
where x = any digits from 0 to 9.
When using the conference rooms (501xxxx) you’ll be asked for a “PIN number”. The first person to enter a ‘room’ may specify a PIN by entering a number (followed by #) to limit access to the conference room, just entering # will make this conference room public. When the last person leaves a conference room, the PIN will be cancelled and others may use the ‘room’.

openbsd98324 · 06-25-2022, 06:58 AM

It seems to play and show something ...

3333@sip2sip.info

openbsd98324 · 06-25-2022, 07:06 AM

Is there a possibility to test if it works?

like audio / video quality. sip/voip rooms would be welcome