Web SDK 2.0 - EN
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Table of Contents

Example of delivery of

...

video stream from SIP to RTMP server with sound

...

injection to the stream

This example demonstrates how to make a call to SIP, receive audio and video traffic from SIP in response, inject sound from file to the received video stream and then redirect the stream to a third-party RTMP server for further broadcasting

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The code of the example

This example is a simple REST client written on JavaScript, available at:

...

where host is WCS server address.

Analyzing the code

To analyze the code get sipget sip-as-rtmp-2.js file version with hash c306c1bbf49bfcbd8e24be927ae95f63b7dbaaba hash ecbadc3 that can be found found here and is availabe to download in build build 2.0.5212.28.2747.

1. Load the test RTMP player on the page for further RTMP stream playback test.

code

...

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themeRDark

...

.

...

2. REST / HTTP queries sending.

code

Sending is done using POST method with ContentType application/json by AJAX query using jQuery framework.

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function sendREST(url, data, successHandler, errorHandler) {
    console.info("url: " + url);
    console.info("data: " + data);
    $.ajax({
        url: url,
        beforeSend: function ( xhr ) {
            xhr.overrideMimeType( "text/plain;" );
        },
        type: 'POST',
        contentType: 'application/json',
        data: data,
        success: (successHandler === undefined) ? handleAjaxSuccess : successHandler,
        error: (errorHandler === undefined) ? handleAjaxError : errorHandler
    });
}

32.  Making Making outgoing call with REST-request /call/startup

code

Call data (RESTCall) are collected from the boxes on page

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    var url = field("restUrl") + "/call/startup";
    callId = generateCallID();
    ...

    var RESTCall = {};
    RESTCall.toStream = field("rtmpStream");
    RESTCall.hasAudio = field("hasAudio");
    RESTCall.hasVideo = field("hasVideo");
    RESTCall.callId = callId;
    RESTCall.sipLogin = field("sipLogin");
    RESTCall.sipAuthenticationName = field("sipAuthenticationName");
    RESTCall.sipPassword = field("sipPassword");
    RESTCall.sipPort = field("sipPort");
    RESTCall.sipDomain = field("sipDomain");
    RESTCall.sipOutboundProxy = field("sipOutboundProxy");
    RESTCall.appKey = field("appKey");
    RESTCall.sipRegisterRequired = field("sipRegisterRequired");

    for (var key in RESTCall) {
        setCookie(key, RESTCall[key]);
    }

    RESTCall.callee = field("callee");

    var data = JSON.stringify(RESTCall);

    sendREST(url, data);
    startCheckCallStatus();

43. Getting the SIP call status with /call/find REST query.

code

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function getStatus() {
    var url = field("restUrl") + "/call/find";
    currentCallId = { callId: callId };
    $("#callTrace").text(callId + " >>> " + field("rtmpUrl"));
    var data = JSON.stringify(currentCallId);
    sendREST(url, data);
}

54. Sending DTMF signal with /call/send_dtmf REST query.

code

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function sendDTMF(value) {
    var url = field("restUrl") + "/call/send_dtmf";
    var data = {};
    data.callId = callId;
    data.dtmf = value;
    data.type = "RFC2833";
    data = JSON.stringify(data);
    sendREST(url, data);
}

65. Re-publishing the SIP call stream to an RTMP server with sound file injecting to the stream by /push/startup REST query

code

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function startRtmpStream() {
    if (!rtmpStreamStarted) {
        rtmpStreamStarted = true;
        var url = field("restUrl") + "/push/startup";
        var RESTObj = {};
        var options = {};
        if ($("#mute").is(':checked')) {
            options.action = "mute";
        } else if ($("#music").is(':checked')) {
            options.action = "sound_on";
            options.soundFile = "sample.wav";
        }
        RESTObj.streamName = field("rtmpStream");
        RESTObj.rtmpUrl = field("rtmpUrl");
        RESTObj.options = options;
        sendREST(url, JSON.stringify(RESTObj), startupRtmpSuccessHandler, startupRtmpErrorHandler);
        sendDataToPlayer();
        startCheckTransponderStatus();
    }
}

76. Getting RTMP stream status with /push/find REST query.

code

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function getTransponderStatus() {
    var url = field("restUrl") + "/push/find";
    var RESTObj = {};
    // By default transponder's stream name will contain prefix "rtmp_"
    RESTObj.streamName = "rtmp_" + field("rtmpStream");
    RESTObj.rtmpUrl = field("rtmpUrl");
    sendREST(url, JSON.stringify(RESTObj), transponderStatusSuccessHandler, transponderStatusErrorHandler);
}

87. Mute/unmute stream sound.

Mute sound with /push/mute mute code

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function mute() {
    if (rtmpStreamStarted) {
        $("#mute").prop('disabled', true);
        var RESTObj = {};
        RESTObj.mediaSessionId = rtmpMediaSessionId;
        var url = field("restUrl") + "/push/mute";
        sendREST(url, JSON.stringify(RESTObj), muteSuccessHandler, muteErrorHandler);
    }
}

Unmute sound with /push/unmute unmute code

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function unmute() {
    if (rtmpStreamStarted) {
        $("#mute").prop('disabled', true);
        var RESTObj = {};
        RESTObj.mediaSessionId = rtmpMediaSessionId;
        var url = field("restUrl") + "/push/unmute";
        sendREST(url, JSON.stringify(RESTObj), muteSuccessHandler, muteErrorHandler);
    }
}

98. Injecting additional sound to RTMP stream.

Injecting sound from file with /push/sound_on on code

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function soundOn() {
    if (rtmpStreamStarted) {
        $("#music").prop('disabled', true);
        var RESTObj = {};
        RESTObj.mediaSessionId = rtmpMediaSessionId;
        RESTObj.soundFile = "sample.wav";
        RESTObj.loop = false;
        var url = field("restUrl") + "/push/sound_on";
        sendREST(url, JSON.stringify(RESTObj), injectSoundSuccessHandler, injectSoundErrorHandler);
    }
}

Stop injecting sound from file with /push/sound_off off code

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function soundOff() {
    if (rtmpStreamStarted) {
        $("#music").prop('disabled', true);
        var RESTObj = {};
        RESTObj.mediaSessionId = rtmpMediaSessionId;
        var url = field("restUrl") + "/push/sound_off";
        sendREST(url, JSON.stringify(RESTObj), injectSoundSuccessHandler, injectSoundErrorHandler);
    }
}

109. Hangup the SIP call with /call/terminate REST query.

code

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function hangup() {
    var url = field("restUrl") + "/call/terminate";
    var currentCallId = { callId: callId };
    var data = JSON.stringify(currentCallId);
    sendREST(url, data);
}

10. RTMP URL displaying on the page to copy to a third party player

code

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function sendDataToPlayer() {
    var host = field("rtmpUrl")
        .replace("localhost", window.location.hostname)
        .replace("127.0.0.1", window.location.hostname);

    var rtmpStreamPrefix = "rtmp_";
    var url = host + "/" + rtmpStreamPrefix + field("rtmpStream");
    $("#player").text(url);
}