Republishing incoming SIP call to a stream¶

Overview¶

WCS 5.2 allows to receive an incoming SIP call from a PBX and republish this call’s stream as RTMP to another server (Wowza for example). Also, another stream published on WCS or captured from mp4 file can be injected into the SIP call, so SIP caller can see and hear the injected stream.

To do this, WCS should be configured with SIP trunks as SIP callee. Then WCS waits for incoming calls from PBX. When SIP call is established, a stream is created from SIP call and republished to target RTMP server. When call is finished, stream is stopped.

Operation flowchart¶

SIP caller makes a call to SIP trunk number on PBX
PBX forwards a call to WCS as callee
WCS receives media data and creates a stream
WCS connects to RTMP server
WCS publishes SIP call stream to RTMP server

How to get this working¶

WCS configuration¶

On WCS side, the following parameter should be set in flashphoner.properties file

sip_add_contact_id=false

Also, SIP trunk should be set up in /usr/local/FlashphonerWebCallServer/conf/sip_trunk.yml file as follows:

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.

By default, RTMP stream name will be rtmp_0123456, where 0123456 is callee number. To remove a prefix, set the following parameter in flashphoner.properties file

rtmp_transponder_stream_name_prefix=

PBX configuration¶

On PBX side, SIP trunk should be set up to redirect SIP calls to WCS. Calls should be redirected to the port defined in /usr/local/FlashphonerWebCallServer/conf/sip_trunk.yml file (40000 in example above).

For example, OpenSIPS can be set up to route calls as follows:

route{
        ...
        #WCS Sip trunk routing, 00 prefix + XX for server number (e.g. WCS1 => 01) + X for trunk number
        if ($rU =~ "^00050[0-9]+$") {
            # WCS5 address and port
            rewritehostport("192.168.1.5:40000");
            route(relay);
        }
}

Testing¶

For test we use:
- WCS server
- PBX server
- Softphone to make a call
- RTMP server
- VLC to play RTMP stream
Open softphone, connect to PBX server, make a call to a callee number defined in PBX SIP trunk setup, for example 001201234
In VLC player open network stream rtmp://rtmp_server:1935/live/rtmp_001201234
Send /call/inject_stream/startup query to inject stream from a local file
Injected file contents is displayed in softphone video window
Send /call/inject_stream/terminate query to stop stream injection

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 interface 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:

customSipMessageListener.java

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.

Warning

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