Can somebody give an ELI5 explanation of SIP? That’s a lot of functionality.
SIP is the control channel and you use it to login to the server (once registered calls to a number will be sent to your program) and control both sending and receiving calls. The actual audio protocol is RTP, typically 20ms of audio is sent or received per RTP packet.
If you've got VoIP service for a landline, it's probably SIP. I think VoLTE might be SIP too? There's extensions for text messaging over SIP as well.
UMTS an similar 2G and 3G protocols could do neat things like have multiple towers compare the signals received from your phone to enable more reliable calling, but this functionality did not survive to make it into cellular's LTE or 5G New Radio (NR) standards.
The whole PBX was running on SIP using Asterisk.
https://www.t-mobile.com/support/plans-features/t-mobile-vid...
I don't think it still works, and/or it requires some strange confluence of events to work.
I developed 3G video phone dating in 2010.
I recall bambuser also used to have a live video streaming over 3G app around that time.
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 midnight and 1am when cell traffic volumes are low.
I was told UK Telco's give voice data priority as this needs to be realtime and then other data has lower levels of priority making it hard to have a reliable stream of data.
FYI, 56K is the minimum standard for voice calls, although with 4G and beyond, its possible to hear the higher bitrate protocols being used on the main mobile networks, as its not so muffled, like someone turning up the high end frequency's on a graphic equaliser, and they are fully down with the 56k protocols.
I dont know what the minimum video protocol bitrate would be, but I did find this which might be useful. > H.264 will only consume around 10 kbps with around 2 fps in 176x144.
When Skype first came out, it was peer to peer, so as a proof of concept to test the new 3G network I did manage to maintain a voice call for an hour which was impressive, but also shows that mobile networks can maintain the mobile data streams if its not encrypted inside a VPN in the UK.
I don't think any of the US networks do that, unless you want to set up your own phone operator :(
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.