We need to develop a SIP server application that will be part of a larger, complex, system.
- The SIP Application will receive incoming calls from originating SIP server
- The calls have to be SENT to terminating SIP server
- When the terminating SIP server gives a CONNECT message the Application has to:
- Not forward the CONNECT to the orinating SIP Server
- Pass the RTP audio from destination to originating sip server as EARLY MEDIA
- No need to send audio from originating to destination sip server, of course.
- After N seconds (N will be in the range 1-5) The Application Will send a CONNECT message to the originating SIP Server and the call will be estabilished in the standard way
In other words, we need to delay the CONNECT message from terminating sip server while passing the one way audio from called party as EARLY MEDIA.
If the solution will be provided using asterisk environment will be preferred.
IP address of destination sip server has to be passed dinamically, call by call.
The application has to be able to handle up to 100 concurrent calls on a standard server hardware.
3 freelance font une offre moyenne de €1033 pour ce travail
I did this before with Asterisk modification. Except I made a sound analysis function, and I delayed 200 OK until I detected human voice, or N second passed. The purpose was to not send false ASR from GSM gateway. Y Plus
Hi, Clarifications needed: 1) What all codecs we want to support for this. 2) How many concurrent calls need to be supported. 3) Do we have a server installed with asterisk. Thanks, Viswanath..