通过任何Asterisk API进行RTP流量访问

Question

I am new to VOIP - please excuse. I am trying to get access to both the actual VOIP SIP header AND RTP traffic using the "asterisk-java" library. I can get access to the SIP header via the FAST AGI, so that is OK and great. Now I want to get access to the RTP traffic once an incoming call has been successfully established, to add additional custom header fields, before passing on in relatively real-time. Question is .... Is this possible using the Asterisk-Java library? - or do I need to delve into the PJSIP library? Please help... Please be gentle.. :-)

Asterisk from source code on linux - could not completely successfully build AND execute without various errors. FreePBX - works OK with asterisk-java library ... Only got as far as using FAST AGI to get SIP header info.

I am after the actual RTP traffic to add additional info.

Answer 1

There are no easy access to sound stream from AGI

You can use UniRTP and conference. Or chan_alsa(sound card), JACK interface etc.

If you want rtp packets(not sound), then you have use libcapture(external) or packet mirroring(see HOMER software)