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Baresip – An Open Source modular SIP User-Agent with audio and video support

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Re: Baresip – An Open Source modular SIP User-Agent with audio and video support

#2
Very nice library. I used it some time ago to implement a module that applied ASR and other audio analytics to audio streams in real time and relayed extracted information via SIP messages. As with many projects in that space, it doesn’t come with a lot of documentation.

Re: Baresip – An Open Source modular SIP User-Agent with audio and video support

#4

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

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.

Re: Baresip – An Open Source modular SIP User-Agent with audio and video support

#5

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

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.

Re: Baresip – An Open Source modular SIP User-Agent with audio and video support

#6
post #4

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

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.

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.

Re: Baresip – An Open Source modular SIP User-Agent with audio and video support

#8
post #4

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

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.

So where does that leave H.323?

Re: Baresip – An Open Source modular SIP User-Agent with audio and video support

#9
post #4

Earlier quoted context omitted.

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.

So where does that leave H.323?

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.

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