Search found 4 matches
- Thu Jul 19, 2007 3:17 am
- Forum: Brekeke SIP Server Forum
- Topic: TLS SRTP Supports and Session Management by external API
- Replies: 4
- Views: 8571
WRT your query regarding session management and call disconnection. This could be achieved using the accounting plugin and dial-plan plugin features provided in V2 upwards. The accounting plugin can tell you when users initiate, start talking (200) and when the session tears down. You could store ...
- Thu Jul 19, 2007 3:10 am
- Forum: Brekeke SIP Server Forum
- Topic: TLS SRTP Supports and Session Management by external API
- Replies: 4
- Views: 8571
- Thu Jul 19, 2007 3:04 am
- Forum: Brekeke SIP Server Forum
- Topic: Hardware and concurrent limit
- Replies: 7
- Views: 11884
Another note, when you are relaying media, consider that the type of media and packet size really matters. You may want to decline media that is G.711 or only allow G.729 / iLBC. You can use &net.rtp.audio.payloadtype in a deploy pattern to force a specific codec. If you do need to allow G.711, I ...
- Thu Jul 19, 2007 2:25 am
- Forum: Brekeke SIP Server Forum
- Topic: Hardware and concurrent limit
- Replies: 7
- Views: 11884
I can't give you any official figures, but can give you some guidance to general capacity and what we experience. Each concurrent dialogue uses one thread for the dialogue, then another thread for each port opened for media relay. This means you can expect five threads to be spawned for each call if ...