Hello!

In my experience and opinion, it will be easier for you to start by reading the book "SIP routing with Kamailio".

But you need to understand that Kamailio is a Proxy and it will be difficult to build a full-fledged RTC platform only on it.

Usually, it is used in conjunction with Asterisk/Freeswitch.

вт, 29 нояб. 2022 г. в 09:10, Clint Crabtree <clint.crabtree@hotmail.com>:

Hello,

 

I’m a DIY entrepreneurial developer beginning to build a system and I stumbled upon Kamailio.   I’ve installed asterisk and kamailio in hopes of setting up an IVR to collect voice to speech information from incoming calls.  I’m skilled with PHP to Javascript application development.  In my research of PBX and discovery of Asterisk I have found enough information online to begin developing with Asterisk ARI; however, somewhere in my discovery process was the suggestion that Kamailio can offer close to real time voice recognition streaming whereas Asterisk is more of a record, pause, translate, respond process. 

 

In an AstriCon presentation by Fred Posner I saw recommendations for new applications the suggestion to develop with Kamailio: https://www.youtube.com/watch?v=IaYOboZQEw0?t=23m10s

 

I’m a newbie to SIP, but it seems really promising, can you help point me in the right direction for aforementioned use-case for Kamailio? or am I misunderstanding some if its capability?

 

Regards,

Clint Crabtree

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BR,

Denys Pozniak

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