This modules will handle all SIP/SDP messages from/to clients, IVR application, and agents, such as: REGISTER, INVITE, BYE, REFER, ACK...
It’s responsible to initialize, manage the sessions, client and transport the messages (SIP/SDP) between clients via UDP sockets.
The main responsibility of server is playback the audio script to the clients base on request from IVR application, and listen DTMF event from clients and return back to IVR application. Media server will make 2 players: one for sending audio data via RTP and one for receiving audio data via RTP. The players are basically gstreamer pipelines.
Tt exists as a SIP client, which can response a call: accept (200 OK), cancel (CANCEL), hang up (BYE) ... However, the different is it can communicate to the media server for playing audio script, which user can listen as a automatically guidance. Beside, IVR application can listen to event from media server corresponding DTMF key event, then request SIP server to redirect (REFER) the call to agents. The communication between IVR, SIP Server and Media Server all via UDP sockets.
Another IVR application which using PJSIP library, it's just a simple app for testing.
- gstreamer 1.16+
- openssl
- cmake 3.16+
git clone git@github.com:humax-embedded-platform/SIP-IVR.gitcd SIP-IVR
source launch.sh -d -m /home/phongdang/WorkSpace/SipServer/blob-d : build with debug mode
-m : your directory that store audio scripts
cd SIP-IVR
mkdir -p build
cd build
cmake ..
make
./sip-ivr -i 192.168.0.3 -m /home/phongdang/WorkSpace/SipServer/blob # correct you media path-i : your server IP address
-m : your directory that store audio scripts
This project tested and supported Zoiper Softphone only, with other one, we hasn't tested yet. Supported media codecs:
- opus/48000
- speex/16000