This is a bit off-topic but I'm wondering if I'm going to have to hack baresip to implement it: Is there such thing as a SIP to mobile phone bridge, so that you can make phone calls over the net via a mobile phone that is in some remote location? The application in my case is making local calls in a country that is expensive to call into. If it exists its the kind of thing that is resistant to search, unless it's cal…
Baresip – An Open Source modular SIP User-Agent with audio and video support
41–50 of 57 posts
Re: Baresip – An Open Source modular SIP User-Agent with audio and video support
#42This is a bit off-topic but I'm wondering if I'm going to have to hack baresip to implement it: Is there such thing as a SIP to mobile phone bridge, so that you can make phone calls over the net via a mobile phone that is in some remote location? The application in my case is making local calls in a country that is expensive to call into. If it exists its the kind of thing that is resistant to search, unless it's cal…
It is one of those slow burn projects I work on every couple of months. mainly because I hate doing anything on the phone. but the idea is I have an asterisk server at the office that ties into the dialed phone world via sip trunk. I should be able to put a sip client on the phone, tunnel a secure connection to the office then the cell phone is just another office phone... anywhere in the world.
Only piece left is the tunnel, I have my heart set on ipsec. is this folly? should I be looking into other tunnel tech?
Re: Baresip – An Open Source modular SIP User-Agent with audio and video support
#43This is a bit off-topic but I'm wondering if I'm going to have to hack baresip to implement it: Is there such thing as a SIP to mobile phone bridge, so that you can make phone calls over the net via a mobile phone that is in some remote location? The application in my case is making local calls in a country that is expensive to call into. If it exists its the kind of thing that is resistant to search, unless it's cal…
Re: Baresip – An Open Source modular SIP User-Agent with audio and video support
#44Re: Baresip – An Open Source modular SIP User-Agent with audio and video support
#45Interesting. I'm trying to (in background mode) build an open source SIP video doorbell, and so far finding a good client that can work with raw video capture devices proved tricky. All the clients with video support need full X-Windows or Wayland stacks, which I most definitely don't want. This seems to be exactly what I need! Now, why did you have to write it in pure C?!? It's not a big deal for me, my VoIP devices…
Re: Baresip – An Open Source modular SIP User-Agent with audio and video support
#46This is a bit off-topic but I'm wondering if I'm going to have to hack baresip to implement it: Is there such thing as a SIP to mobile phone bridge, so that you can make phone calls over the net via a mobile phone that is in some remote location? The application in my case is making local calls in a country that is expensive to call into. If it exists its the kind of thing that is resistant to search, unless it's cal…
I am using baresip on my phone(from fdroid). I don't have my whole stack plumbed yet so I can't say how well it works. but it connects to my asterisk server via wifi well enough. I was expecting sip to be built into android but apparently it was removed in 10 or 11. but found the baresip package. I like baresip well enough for desktop use so I decided to try it on the phone. It is one of those slow burn projects I wo…
Getting just plain SIP and RTP through consumer, enterprise, and cell carrier nets can be troublesome. For example, SIP especially is NAT-antagonistic, and edge routers can try to be "helpful" and do half-baked proxying or deep packet mangling that just messes up your smarter VoIP server. Consumer ISPs have a tendency to block SIP altogether. Throwing in encryption complications may give you another layer of hard-to-diagnose headaches. (Of course, if you can get the secure layer through the net it might force intermediaries to keep their fingers out. Other tricks like using non-standard ports can help with that, too.)
Source: used to run the signalling/media dev team for a VoIP provder many moons ago, but we didn't bother with the secure modes/protocols.
[1] https://en.wikipedia.org/wiki/Session_Initiation_Protocol#En...
[2] https://en.wikipedia.org/wiki/Secure_Real-time_Transport_Pro...
Re: Baresip – An Open Source modular SIP User-Agent with audio and video support
#47Can somebody give an ELI5 explanation of SIP? That’s a lot of functionality.
As for the complications... Yeah.
The folks who invented VoIP had a dream. They saw the existing extremely centralized, extremely locked-down phone carriers as The Enemy. Instead, they envisioned a loose decentralized federation of peers who directly contacted one another over the Internet to make calls.
Since there weren't supposed to be any all-knowing, all-powerful, all-standard-enforcing choke points, everything had to be negotiated and a consensus built up across the interconnected parts as to the call state and control. Worse, intermediate services (proxies) were allowed to be stateless. So the consensus has to be continually reinforced during and after the call.
Another design choice was to split functionality into lots of optional tiny pieces. "SIP" has a bazillion RFCs (standards specs), and some pretty byzantine control flows to implement all the interconnections. (To be fair, The Enemy is just as bad or worse. Both camps were dealing with extremely limited computing power and bandwidth in the distant past, plus incremental feature enhancement.)
My poster child is "call parking". This allows you to "park" a call from one phone by putting it on hold, and then picking it on another. The suggested SIP call from (don't remember which RFC it's part of) involves specialized "parking services"; all these do is receive a call and then later forward it. The minimal flow diagram has dozens of distinct steps involving half-a-dozen players.
SIP was a standard, not proprietary, and "good enough", so became the overwhelming standard for interconnecton of VoIP players. In reality, the vast majority of conrol flows are:
1. Phone says "hey central server, here's a phone number. Call them."
2. Central server says "Hey phone, here's an incoming call."
3. Phone says: "OK, I'll answer".
4. Phone to server: "Hang up".
5. Server to phone: "Hang up".
6. Phone: "Here's some random buttons my user pushed mid-call." (This includes digits for IVR menu trees, and on/off hold).
Audio almost always uses one of three or four popular encoding choices: 1) "Dead stupid" (G.711), 2) "Bad compression" (G.729), 3) "Oooh! Hifi!" (G.722), 4) "Something open, decent, and from the current century" (Opus). Video is usually H.263 or one of its relatives.
Re: Baresip – An Open Source modular SIP User-Agent with audio and video support
#48Interesting. I'm trying to (in background mode) build an open source SIP video doorbell, and so far finding a good client that can work with raw video capture devices proved tricky. All the clients with video support need full X-Windows or Wayland stacks, which I most definitely don't want. This seems to be exactly what I need! Now, why did you have to write it in pure C?!? It's not a big deal for me, my VoIP devices…
If you're just receiving and sending video and a bell signal to/from static IP, What problem is SIP solving? I mean can't you just put up gstreamer to take video from an esp32cam?
Re: Baresip – An Open Source modular SIP User-Agent with audio and video support
#49Earlier quoted context omitted.
I'm really intrigued. Is it possible to get video calling working with mobile networks' own video calling support over LTE? I could never find how it's done.
I've not done that personally, but I'd have a look on the freeswitch forums to see if someone on there has already done it. I have had the free 3CX sip client on an android mobile, using a UDP vpn back to a freeswitch server which was connected to the UK phone system. Proof of concept thing. Because of the nature of the mobile data, which is like bursts of data, it made it hard to have a conversation even around midn…
I don't think any of the US networks do that, unless you want to set up your own phone operator :(
Re: Baresip – An Open Source modular SIP User-Agent with audio and video support
#50Earlier quoted context omitted.
It's not so much a threat model, but maintainability and reliability. I want to have as few dependencies as possible, and for the code to work with minimal to no changes for the next 10 years. Ideally, I would use something like HomeAssistant Operating System for the base, with my code running in a Docker container.
If it's about maintainability and reliability, and not so much a threat model, why does the fact that your VoIP devices are already in a separate VLAN without external access mean it's not a big deal for you? > I want to have as few dependencies as possible, and for the code to work with minimal to no changes for the next 10 years. Then a C program sounds like an ideal choice!
VLAN is what makes it not such a big deal. However, I want to offer the same solution to other people in our HOA, and I most definitely don't want to set up VLANs for each one of them.
> Then a C program sounds like an ideal choice!
Not really? No package management, for one thing.