baresip
Baresip is a modular SIP User-Agent with audio and video support (by baresip)
waterfall
A cascading stream forwarding unit for scalable, distributed voice and video conferencing over Matrix (by matrix-org)
baresip | waterfall | |
---|---|---|
7 | 1 | |
1,618 | 95 | |
1.0% | - | |
9.4 | 5.5 | |
7 days ago | about 1 year ago | |
C | Go | |
BSD 3-clause "New" or "Revised" License | Apache License 2.0 |
The number of mentions indicates the total number of mentions that we've tracked plus the number of user suggested alternatives.
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.
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
Posts with mentions or reviews of baresip.
We have used some of these posts to build our list of alternatives
and similar projects. The last one was on 2023-09-02.
- 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
waterfall
Posts with mentions or reviews of waterfall.
We have used some of these posts to build our list of alternatives
and similar projects. The last one was on 2022-06-10.
-
20 Years of Sip – A Retrospective
It is a shame. I don't see any easy answer. I am working on sfu-to-sfu[0] and hope it can make some traction. If we can get all the Open Source WebRTC servers working together, maybe there is hope? I believe WebRTC did the right thing. It was as flexible as possible to make it more palatable. You can standardized/codify things, but you can't undo it :)
I am also really excited about WHIP[1]
[0] https://github.com/matrix-org/sfu-to-sfu
[1] https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
What are some alternatives?
When comparing baresip and waterfall you can also consider the following projects:
tSIP - SIP softphone
pjproject - PJSIP project