baresip
freeswitch
baresip | freeswitch | |
---|---|---|
7 | 3 | |
1,618 | 3,131 | |
1.0% | 1.7% | |
9.4 | 8.6 | |
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
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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.
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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
freeswitch
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Virtual line to offer audiobooks by phone?
Hopefully an update to allow speed changes will land... https://github.com/signalwire/freeswitch/pull/244
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Ask HN: Did Plivo silently abandon open sourcing their software?
Some clarification on what you mean would be great.
When I look at their organization on Github I see quite a few active projects e.g. "Sharq" - SHARQ Server is an flexible, rate limited queuing system based on the SHARQ Core library and Redis. (https://github.com/plivo/sharq-server) as well as client packages in quite a few languages.
Beyond that the documentation you referenced clearly points out that FreeSwitch (https://github.com/signalwire/freeswitch) is the telephony engine (https://web.archive.org/web/20130314113846/http://docs.plivo...) and the install script from the docs shows that their roots are in Pilvo being a Python web app (https://github.com/plivo/plivoframework/blob/master/scripts/...). I also see a few Django applications in their repositories as well.
Thus, they open sourced their software. Are they open sourcing all of their software? Likely not. It could be that they open sourced much of their code and then got no contributions.
If they weren't receiving any contributions in alignment with their development, there would be no value for them to continue to have that distraction from running their business. It could also be that they didn't have the resources to _maintain_ external facing repositories, issues, and the communities around them.
Providing a more clear explanation of your expectations would help get you the best answer.
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UDM Pro listening on undocumented TCP port 2856 with cert valid for 100 years - what is this and how do I turn it off?
FreeSwitch is a telecom stack similar to Asterisk. See here: freeswitch. It would be to support VoIP on the UDMP.
What are some alternatives?
tSIP - SIP softphone
Kamailio - Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
pjproject - PJSIP project
libresbc - An open source Session Border Controller 🌟 The SBC you dream about 🗽 LibreSBC will help you save thousands of dollars.
waterfall - A cascading stream forwarding unit for scalable, distributed voice and video conferencing over Matrix
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.
ASTPP - Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://jira.astppbilling.org
pade - Pà dé (Yoruba word for Meet) is a browser extension (Chrome/Edge) based unified communications desktop client for Openfire.
PHP SIP Parsing/Rendering Library - :phone: SIP Parsing/Rendering Library for PHP
WebphoneLib - Easier web calling by providing a layer of abstraction around SIP.js
EEGwithRaspberryPI - Open-Source board for converting RaspberryPI to Brain-computer interface [Moved to: https://github.com/HackerBCI/EEGwithRaspberryPI]