5 ms·

Can somebody give an ELI5 explanation of SIP? That’s a lot of functionality.

toast0 3y ago
SIP is more or less the signaling protocol for telephone service as VoIP. SIP doesn't include the actual audio, which is usually RTP on a different socket, but this client seems to do audio too. 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.
- simfree 3y ago
  VoLTE and VoNR are just SIP over a cell carrier prioritized data channel (Google LTE QCI). T-Mobile has a beta for latency sensitive applications to access the 5G equivalent of this, it's called Network Slicing. 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.
- doubleg72 3y ago
  So where does that leave H.323?
  - alophawen 3y ago
    Last I worked with telephony (around 2010), IIRC we recieved calls over 3G network as SS7/H323 using Dialogic hardware and converted them on the fly to SIP as they entered our systems. The whole PBX was running on SIP using Asterisk.
    - cyberax 3y ago
      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.
      - posguy 3y ago
        Yes, T-Mobile and a few other carriers support videocalling right in the Android dialer. https://www.t-mobile.com/support/plans-features/t-mobile-video-calling https://www.t-mobile.com/support/plans-features/t-mobile-vid...
        cyberax 3y ago
        I've seen it work _once_, around 8 years ago on a T-Mobile branded Galaxy S phone. I don't think it still works, and/or it requires some strange confluence of events to work.
      - alophawen 3y ago
        I managed to get video working between 3G and SIP, so I expect so. 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.
      - darkclouds 3y ago
        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 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.
  - yed 3y ago
    It’s mostly a legacy protocol at this point.
    - doubleg72 3y ago
      Just for the record, that isn’t true. H.323 is a VoIP protocol and is implemented on a lot of private PBX systems. Most on Prem systems can handle SIP or H.323, likewise for about any standard enterprise IP phone. H.323 works out of the box, pure SIP requires additional infrastructure, mainly SBC or session boarder controller.
shrubble 3y ago
SIP is a protocol for Voice over Internet Protocol or VOIP. 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.
userbinator 3y ago
The open interoperable standard for voice and video calls.
mgsouth 3y ago
As the other comments have noted, it's the "control plane" that tells all the Voice-over-IP parts how to set up and tear down a call. 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.

SIP is more or less the signaling protocol for telephone service as VoIP. SIP doesn't include the actual audio, which is usually RTP on a different socket, but this client seems to do audio too. 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.

simfree 3y ago
VoLTE and VoNR are just SIP over a cell carrier prioritized data channel (Google LTE QCI). T-Mobile has a beta for latency sensitive applications to access the 5G equivalent of this, it's called Network Slicing. 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.
doubleg72 3y ago
So where does that leave H.323?
- alophawen 3y ago
  Last I worked with telephony (around 2010), IIRC we recieved calls over 3G network as SS7/H323 using Dialogic hardware and converted them on the fly to SIP as they entered our systems. The whole PBX was running on SIP using Asterisk.
  - cyberax 3y ago
    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.
    - posguy 3y ago
      Yes, T-Mobile and a few other carriers support videocalling right in the Android dialer. https://www.t-mobile.com/support/plans-features/t-mobile-video-calling https://www.t-mobile.com/support/plans-features/t-mobile-vid...
      - cyberax 3y ago
        I've seen it work _once_, around 8 years ago on a T-Mobile branded Galaxy S phone. I don't think it still works, and/or it requires some strange confluence of events to work.
    - alophawen 3y ago
      I managed to get video working between 3G and SIP, so I expect so. 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.
    - darkclouds 3y ago
      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 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.
- yed 3y ago
  It’s mostly a legacy protocol at this point.
  - doubleg72 3y ago
    Just for the record, that isn’t true. H.323 is a VoIP protocol and is implemented on a lot of private PBX systems. Most on Prem systems can handle SIP or H.323, likewise for about any standard enterprise IP phone. H.323 works out of the box, pure SIP requires additional infrastructure, mainly SBC or session boarder controller.

shrubble 3y ago

SIP is a protocol for Voice over Internet Protocol or VOIP. 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.

userbinator 3y ago

The open interoperable standard for voice and video calls.

mgsouth 3y ago

As the other comments have noted, it's the "control plane" that tells all the Voice-over-IP parts how to set up and tear down a call. 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.