...

trunks:
 pbx_t0:
  localPort : 40000
  proxyIp : pbx_address
  remotePort : 5060
  url : rtmp://rtmp_server:1935/live
  visibleName : CUSTOM_NAME
  sdp : |
   v=0
   o=10009 2469 1555 IN IP4 0.0.0.0
   c=IN IP4 0.0.0.0
   t=0 0
   m=audio 7270 RTP/AVP 96
   a=rtpmap:96 opus/48000/2
   a=recvonly
   m=video 9202 RTP/AVP 96
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=42801F
   a=recvonly
  sdpParams :
   - b=AS:2000
   - b=RS:50
   - b=RR:100

Where

• pbx_t0 – WCS SIP trunk name
• localPort – port to receive incoming SIP calls
• proxyIp – PBX address
• remotePort – PBX port to register
• url – RTMP server URL to republish SIP call stream
• visibleName - name to display to caller, this name is sending to PBX when register
• sdp – SDP to send to PBX in 200 OK response
• sdpParams - parameter to insert to SDP for SIP call media bitrate and channel bandwidth management

WCS supports both TCP and UDP transport for SIP calls, so it listens for incoming both TCP and UDP port defined by localPort parameter.

...

3. Send /call/inject_stream/startup query to inject stream from a local file

Image RemovedImage Added

4. Injected file contents is displayed in softphone video window

...

5. Send /call/inject_stream/terminate query to stop stream injection

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Custom SIP messages listener implementation

In some cases, additional handling of incoming SIP messages is required. To do this, a custom Java class implementing ISipMessageListener should be developed to catch incoming SIP messages and handle them as needed.

Look at the example to add port to Request URI of INVITE request if port is not set by callee. Here is the class source code:

package com.customListener;

import com.flashphoner.sdk.sip.ISipMessageListener;
import com.flashphoner.server.client.IClient;
import gov.nist.javax.sip.address.Authority;
import gov.nist.javax.sip.address.SipUri;
import gov.nist.javax.sip.message.SIPMessage;
import gov.nist.javax.sip.message.SIPRequest;
import gov.nist.javax.sip.stack.MessageChannel;
import org.slf4j.Logger;
import org.slf4j.LoggerFactory;

import javax.sip.message.Request;

public class customSipMessageListener implements ISipMessageListener {

    private static Logger log = LoggerFactory.getLogger("customSipMessageListener");

    @Override
    public void processMessage(SIPMessage sipMessage, IClient client, MessageChannel channel) {
        if (sipMessage instanceof SIPRequest) {
            SIPRequest request = (SIPRequest) sipMessage;
            String method = request.getRequestLine().getMethod();
            if (Request.INVITE.equals(method)) {
                Authority authority = ((SipUri)request.getRequestURI()).getAuthority();
                int port = authority.getPort();
                if (port <= 0) {
		            if (log.isDebugEnabled()) {
			             log.debug("Inject port " + channel.getPort());
		            }
                    authority.setPort(channel.getPort());
                }
            }
        }
    }
}

Make the folder structure in home directory on server

mkdir -p com/customListener

Copy the class source code to the folder created and compile it

javac -cp "/usr/local/FlashphonerWebCallServer/lib/*" ./com/customListener/customSipMessageListener.java

Pack the class compiled to the jar file

jar cf customSipMessageListener.jar com/customListener/customSipMessageListener.class

Copy jar file to server folder

cp customSipMessageListener.jar /usr/local/FlashphonerWebCallServer/lib

It is necessary to set the class developed to the following parameter in flashphoner.properties file

sip_msg_listener=com.customListener.customSipMessageListener

then restart server.

Incoming SIP call stream recording

Incoming SIP call streams can be recorded on server. Set the following parameter in flashphoner.properties file to record all the incoming SIP calls:

sip_record_stream=true

Use REST API query to record a certain SIP call stream.

Note that incoming SIP calls will not be recorded if the following setting is active

sip_single_route_only=true

Known issues

1. RTMP stream republished from incoming SIP call can be out of sync

Symptoms: SIP call stream is out of sync while playing it from RTMP server

Solution:

a) set the following parameter

sip_force_rtcp_feedback=true

b) minimize or exclude packet losses in channel between PBX and WCS