Kamailio® (successor of former OpenSER and SER) is an Open Source SIP Server released under GPL, able to handle thousands of call setups per second. Kamailio can be used to build large platforms for VoIP and realtime communications – presence, WebRTC, Instant messaging and other applications. Moreover, it can be easily used for scaling up SIP-to-PSTN gateways, PBX systems or media servers like Asterisk™, FreeSWITCH™ or SEMS.
|Kamailio||2N IP Intercom||Registraion||Secure RTP (SRTP)|
Add new user in kamailio using kamctl tool. In following example firstname.lastname@example.org is number of extension on kamailio with IP address 10.27.1.66 and 5000 will be a password to this extension.
Output should look like following.
- Fill in Extension number in Phone (ID) and Authentication ID from 3CX PBX to Authentication ID.
- Fill in the IP address of kamailio in to Domain, Proxy Address and Registrar address boxes.
- Fill in your Authentication password in to the corresponding box.
SRTP on 3CX Phone System
No special settings are required to be made in 3CX Phone System.
SRTP on 2N IP Intercom
Go to SIP settings in Services and in Advanced settings enable recieving and transmitting of SRTP calls.
- Verified with video
- Verified with audio only
- Work with limitation