baresip
Kamailio
| baresip | Kamailio | |
|---|---|---|
| 7 | 1 | |
| 1,618 | 2,135 | |
| 1.0% | 1.4% | |
| 9.4 | 9.9 | |
| 7 days ago | 2 days ago | |
| C | C | |
| BSD 3-clause "New" or "Revised" License | GNU General Public License v3.0 or later |
Stars - the number of stars that a project has on GitHub. Growth - month over month growth in stars.
Activity is a relative number indicating how actively a project is being developed. Recent commits have higher weight than older ones.
For example, an activity of 9.0 indicates that a project is amongst the top 10% of the most actively developed projects that we are tracking.
baresip
- Baresip – An Open Source modular SIP User-Agent with audio and video support
-
Connecting linphone softphone
I found this info on Reals on Linphone but could not get it to work with the strings there either: https://github.com/baresip/baresip/issues/383
- Decent linux softphone w/command-line or dbus?
- High bit rate recording sometimes causes this (I haven't noticed this with AVC encoding, only HEVC), is this normal? GPU: RX 590 / 22.3.1 drivers
- Not able to play audio from a voip daemon program that runs at boot.
-
20 Years of Sip – A Retrospective
Happy SIP user for nearly twenty years, which allows me to bridge three countries. Currently using baresip [1] and finding it to be remarkably reliable, but is there any hardware phone out there that I can put on my desk? Or is the sane thing to do to get a handset and hook it up to a computer via say USB? I have tried at least twice over the years to gain some clarity on these questions, but maybe I am using the wrong search terms?
[1]: https://github.com/baresip/baresip
Kamailio
-
What's the use of fd passing?
Multi-process server applications using a prefork model are one example. You might have one process dedicated to accepting new connections over the network and maybe doing an initial handshake, and then a bunch of worker processes handling the actual connections. Kamailio's CDP module is built like this (https://github.com/kamailio/kamailio/tree/master/src/modules/cdp)
What are some alternatives?
tSIP - SIP softphone
Asterisk - The official Asterisk Project repository.
pjproject - PJSIP project
freeswitch - FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a Raspberry PI to a multi-core server, FreeSWITCH can unlock the telecommunications potential of any device.
waterfall - A cascading stream forwarding unit for scalable, distributed voice and video conferencing over Matrix
Homer - HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
Routr - ⚡ The future of programmable SIP servers.
ASTPP - Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://jira.astppbilling.org
Ostel
SipXcom - Unified Communications System
chan-sccp - Replacement for the SCCP channel driver in Asterisk. Extended features include Shared Lines, Presence / BLF, customizable Feature Buttons, and Custom Device State. Visit our discussion mailing list for help and join us as a developer if you like.
PHP SIP Parsing/Rendering Library - :phone: SIP Parsing/Rendering Library for PHP