/* Autogenerated with Kurento Idl */
/*
* (C) Copyright 2013-2015 Kurento (http://kurento.org/)
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
var inherits = require('inherits');
var kurentoClient = require('kurento-client');
var disguise = kurentoClient.disguise;
var checkType = kurentoClient.checkType;
var ChecktypeError = checkType.ChecktypeError;
var Transaction = kurentoClient.TransactionsManager.Transaction;
var SdpEndpoint = require('./SdpEndpoint');
function noop(error, result) {
if (error) console.trace(error);
return result
};
/**
* @classdesc
* Handles RTP communications.
* <p>
* All endpoints that rely on the RTP protocol, like the
* <strong>RtpEndpoint</strong> or the <strong>WebRtcEndpoint</strong>,
* inherit
* from this class. The endpoint provides information about the connection
* state
* and the media state, which can be consulted at any time through the
* {@link module:core/abstracts.BaseRtpEndpoint#mediaState} and the {@link
* module:core/abstracts.BaseRtpEndpoint#connectionState} properties. It is
* also possible subscribe to events fired when these properties change.
* </p>
* <ul>
* <li>
* ConnectionStateChangedEvent: This event is raised when the connection
* between two peers changes. It can have two values:
* <ul>
* <li>CONNECTED</li>
* <li>DISCONNECTED</li>
* </ul>
* </li>
* <li>
* MediaStateChangedEvent: Based on RTCP packet flow, this event provides
* more
* reliable information about the state of media flow. Since RTCP packets
* are
* not flowing at a constant rate (minimizing a browser with an
* RTCPeerConnection might affect this interval, for instance), there is a
* guard period of about 5s. This traduces in a period where there might be
* media flowing, but the event hasn't been fired yet. Nevertheless, this
* is
* the most reliable and useful way of knowing what the state of media
* exchange
* is. Possible values are:
* <ul>
* <li>CONNECTED: There is an RTCP packet flow between peers.</li>
* <li>
* DISCONNECTED: No RTCP packets have been received, or at least 5s
* have
* passed since the last packet arrived.
* </li>
* </ul>
* </li>
* </ul>
* <p>
* Part of the bandwidth control of the video component of the media session
* is
* done here:
* </p>
* <ul>
* <li>
* Input bandwidth: Configuration value used to inform remote peers about
* the
* bitrate that can be pushed into this endpoint.
* <ul>
* <li>
* <strong>{get,set}MinVideoRecvBandwidth</strong>: Minimum bitrate
* requested on the received video stream.
* </li>
* <li>
* <strong>{get,set}Max{Audio,Video}RecvBandwidth</strong>: Maximum
* bitrate
* expected for the received stream.
* </li>
* </ul>
* </li>
* <li>
* Output bandwidth: Configuration values used to control bitrate of the
* output
* video stream sent to remote peers. It is important to keep in mind that
* pushed bitrate depends on network and remote peer capabilities. Remote
* peers
* can also announce bandwidth limitation in their SDPs (through the
* <code>b={modifier}:{value}</code> tag). Kurento will always enforce
* bitrate
* limitations specified by the remote peer over internal configurations.
* <ul>
* <li>
* <strong>{get,set}MinVideoSendBandwidth</strong>: Minimum video
* bitrate
* sent to remote peer.
* </li>
* <li>
* <strong>{get,set}MaxVideoSendBandwidth</strong>: Maximum video
* bitrate
* sent to remote peer.
* </li>
* <li>
* <strong>RembParams.rembOnConnect</strong>: Initial local REMB
* bandwidth
* estimation that gets propagated when a new endpoint is connected.
* </li>
* </ul>
* </li>
* </ul>
* <p>
* <strong>
* All bandwidth control parameters must be changed before the SDP
* negotiation
* takes place, and can't be changed afterwards.
* </strong>
* </p>
*
* @abstract
* @extends module:core/abstracts.SdpEndpoint
*
* @constructor module:core/abstracts.BaseRtpEndpoint
*
* @fires {@link module:core#event:ConnectionStateChanged ConnectionStateChanged}
* @fires {@link module:core#event:MediaStateChanged MediaStateChanged}
*/
function BaseRtpEndpoint(){
BaseRtpEndpoint.super_.call(this);
};
inherits(BaseRtpEndpoint, SdpEndpoint);
//
// Public properties
//
/**
* Connection state.
* <ul>
* <li>CONNECTED</li>
* <li>DISCONNECTED</li>
* </ul>
*
* @alias module:core/abstracts.BaseRtpEndpoint#getConnectionState
*
* @param {module:core/abstracts.BaseRtpEndpoint~getConnectionStateCallback} [callback]
*
* @return {external:Promise}
*/
BaseRtpEndpoint.prototype.getConnectionState = function(callback){
var transaction = (arguments[0] instanceof Transaction)
? Array.prototype.shift.apply(arguments)
: undefined;
var usePromise = false;
if (callback == undefined) {
usePromise = true;
}
if(!arguments.length) callback = undefined;
callback = (callback || noop).bind(this)
return disguise(this._invoke(transaction, 'getConnectionState', callback), this)
};
/**
* @callback module:core/abstracts.BaseRtpEndpoint~getConnectionStateCallback
* @param {external:Error} error
* @param {module:core/complexTypes.ConnectionState} result
*/
/**
* Maximum video bitrate sent to remote peer.
* <p>
* With this parameter you can control the maximum video quality that will be
* sent when reacting to good network conditions. Setting this parameter to a
* high value permits the video quality to raise when the network conditions
* get
* better.
* </p>
* <p>
* This parameter provides a way to limit the bitrate requested by remote REMB
* bandwidth estimations: the bitrate sent will be always equal or less than
* this parameter, even if the remote peer requests higher bitrates.
* </p>
* <p>
* Note that the default value of <strong>500 kbps</strong> is a VERY
* conservative one, and leads to a low maximum video quality. Most
* applications
* will probably want to increase this parameter to higher values such as 2000
* mbps) or even 10000 (10 mbps).
* </p>
* <p>
* The REMB congestion control algorithm works by gradually increasing the
* output
* video bitrate, until the available bandwidth is fully used or the maximum
* send
* bitrate has been reached. This is a slow, progressive change, which starts
* at
* 300 kbps by default. You can change the default starting point of REMB
* estimations, by setting <code>RembParams.rembOnConnect</code>.
* </p>
* <ul>
* <li>Unit: kbps (kilobits per second).</li>
* <li>Default: 500.</li>
* <li>
* 0 = unconstrained: the video bitrate will grow until all the available
* network bandwidth is used by the stream.<br />
* Note that this might have a bad effect if more than one stream is running
* (as all of them would try to raise the video bitrate indefinitely, until
* the
* network gets saturated).
* </li>
* </ul>
*
* @alias module:core/abstracts.BaseRtpEndpoint#getMaxVideoSendBandwidth
*
* @param {module:core/abstracts.BaseRtpEndpoint~getMaxVideoSendBandwidthCallback} [callback]
*
* @return {external:Promise}
*/
BaseRtpEndpoint.prototype.getMaxVideoSendBandwidth = function(callback){
var transaction = (arguments[0] instanceof Transaction)
? Array.prototype.shift.apply(arguments)
: undefined;
var usePromise = false;
if (callback == undefined) {
usePromise = true;
}
if(!arguments.length) callback = undefined;
callback = (callback || noop).bind(this)
return disguise(this._invoke(transaction, 'getMaxVideoSendBandwidth', callback), this)
};
/**
* @callback module:core/abstracts.BaseRtpEndpoint~getMaxVideoSendBandwidthCallback
* @param {external:Error} error
* @param {external:Integer} result
*/
/**
* Maximum video bitrate sent to remote peer.
* <p>
* With this parameter you can control the maximum video quality that will be
* sent when reacting to good network conditions. Setting this parameter to a
* high value permits the video quality to raise when the network conditions
* get
* better.
* </p>
* <p>
* This parameter provides a way to limit the bitrate requested by remote REMB
* bandwidth estimations: the bitrate sent will be always equal or less than
* this parameter, even if the remote peer requests higher bitrates.
* </p>
* <p>
* Note that the default value of <strong>500 kbps</strong> is a VERY
* conservative one, and leads to a low maximum video quality. Most
* applications
* will probably want to increase this parameter to higher values such as 2000
* mbps) or even 10000 (10 mbps).
* </p>
* <p>
* The REMB congestion control algorithm works by gradually increasing the
* output
* video bitrate, until the available bandwidth is fully used or the maximum
* send
* bitrate has been reached. This is a slow, progressive change, which starts
* at
* 300 kbps by default. You can change the default starting point of REMB
* estimations, by setting <code>RembParams.rembOnConnect</code>.
* </p>
* <ul>
* <li>Unit: kbps (kilobits per second).</li>
* <li>Default: 500.</li>
* <li>
* 0 = unconstrained: the video bitrate will grow until all the available
* network bandwidth is used by the stream.<br />
* Note that this might have a bad effect if more than one stream is running
* (as all of them would try to raise the video bitrate indefinitely, until
* the
* network gets saturated).
* </li>
* </ul>
*
* @alias module:core/abstracts.BaseRtpEndpoint#setMaxVideoSendBandwidth
*
* @param {external:Integer} maxVideoSendBandwidth
* @param {module:core/abstracts.BaseRtpEndpoint~setMaxVideoSendBandwidthCallback} [callback]
*
* @return {external:Promise}
*/
BaseRtpEndpoint.prototype.setMaxVideoSendBandwidth = function(maxVideoSendBandwidth, callback){
var transaction = (arguments[0] instanceof Transaction)
? Array.prototype.shift.apply(arguments)
: undefined;
//
// checkType('int', 'maxVideoSendBandwidth', maxVideoSendBandwidth, {required: true});
//
var params = {
maxVideoSendBandwidth: maxVideoSendBandwidth
};
callback = (callback || noop).bind(this)
return disguise(this._invoke(transaction, 'setMaxVideoSendBandwidth', params, callback), this)
};
/**
* @callback module:core/abstracts.BaseRtpEndpoint~setMaxVideoSendBandwidthCallback
* @param {external:Error} error
*/
/**
* Media flow state.
* <ul>
* <li>CONNECTED: There is an RTCP flow.</li>
* <li>DISCONNECTED: No RTCP packets have been received for at least 5
* sec.</li>
* </ul>
*
* @alias module:core/abstracts.BaseRtpEndpoint#getMediaState
*
* @param {module:core/abstracts.BaseRtpEndpoint~getMediaStateCallback} [callback]
*
* @return {external:Promise}
*/
BaseRtpEndpoint.prototype.getMediaState = function(callback){
var transaction = (arguments[0] instanceof Transaction)
? Array.prototype.shift.apply(arguments)
: undefined;
var usePromise = false;
if (callback == undefined) {
usePromise = true;
}
if(!arguments.length) callback = undefined;
callback = (callback || noop).bind(this)
return disguise(this._invoke(transaction, 'getMediaState', callback), this)
};
/**
* @callback module:core/abstracts.BaseRtpEndpoint~getMediaStateCallback
* @param {external:Error} error
* @param {module:core/complexTypes.MediaState} result
*/
/**
* Minimum bitrate requested on the received video stream.
* <p>
* This is used to set a minimum value of local REMB during bandwidth
* estimation,
* if supported by the implementing class. The REMB estimation will then be
* sent
* to remote peers, requesting them to send at least the indicated video
* bitrate.
* It follows that min values will only have effect in remote peers that
* support
* this congestion control mechanism, such as Chrome.
* </p>
* <ul>
* <li>Unit: kbps (kilobits per second).</li>
* <li>Default: 0.</li>
* <li>
* Note: The absolute minimum REMB value is 30 kbps, even if a lower value
* is
* set here.
* </li>
* </ul>
*
* @alias module:core/abstracts.BaseRtpEndpoint#getMinVideoRecvBandwidth
*
* @param {module:core/abstracts.BaseRtpEndpoint~getMinVideoRecvBandwidthCallback} [callback]
*
* @return {external:Promise}
*/
BaseRtpEndpoint.prototype.getMinVideoRecvBandwidth = function(callback){
var transaction = (arguments[0] instanceof Transaction)
? Array.prototype.shift.apply(arguments)
: undefined;
var usePromise = false;
if (callback == undefined) {
usePromise = true;
}
if(!arguments.length) callback = undefined;
callback = (callback || noop).bind(this)
return disguise(this._invoke(transaction, 'getMinVideoRecvBandwidth', callback), this)
};
/**
* @callback module:core/abstracts.BaseRtpEndpoint~getMinVideoRecvBandwidthCallback
* @param {external:Error} error
* @param {external:Integer} result
*/
/**
* Minimum bitrate requested on the received video stream.
* <p>
* This is used to set a minimum value of local REMB during bandwidth
* estimation,
* if supported by the implementing class. The REMB estimation will then be
* sent
* to remote peers, requesting them to send at least the indicated video
* bitrate.
* It follows that min values will only have effect in remote peers that
* support
* this congestion control mechanism, such as Chrome.
* </p>
* <ul>
* <li>Unit: kbps (kilobits per second).</li>
* <li>Default: 0.</li>
* <li>
* Note: The absolute minimum REMB value is 30 kbps, even if a lower value
* is
* set here.
* </li>
* </ul>
*
* @alias module:core/abstracts.BaseRtpEndpoint#setMinVideoRecvBandwidth
*
* @param {external:Integer} minVideoRecvBandwidth
* @param {module:core/abstracts.BaseRtpEndpoint~setMinVideoRecvBandwidthCallback} [callback]
*
* @return {external:Promise}
*/
BaseRtpEndpoint.prototype.setMinVideoRecvBandwidth = function(minVideoRecvBandwidth, callback){
var transaction = (arguments[0] instanceof Transaction)
? Array.prototype.shift.apply(arguments)
: undefined;
//
// checkType('int', 'minVideoRecvBandwidth', minVideoRecvBandwidth, {required: true});
//
var params = {
minVideoRecvBandwidth: minVideoRecvBandwidth
};
callback = (callback || noop).bind(this)
return disguise(this._invoke(transaction, 'setMinVideoRecvBandwidth', params, callback), this)
};
/**
* @callback module:core/abstracts.BaseRtpEndpoint~setMinVideoRecvBandwidthCallback
* @param {external:Error} error
*/
/**
* Minimum video bitrate sent to remote peer.
* <p>
* With this parameter you can control the minimum video quality that will be
* sent when reacting to bad network conditions. Setting this parameter to a
* low
* value permits the video quality to drop when the network conditions get
* worse.
* </p>
* <p>
* This parameter provides a way to override the bitrate requested by remote
* REMB
* bandwidth estimations: the bitrate sent will be always equal or greater
* than
* this parameter, even if the remote peer requests even lower bitrates.
* </p>
* <p>
* Note that if you set this parameter too high (trying to avoid bad video
* quality altogether), you would be limiting the adaptation ability of the
* congestion control algorithm, and your stream might be unable to ever
* recover
* from adverse network conditions.
* </p>
* <ul>
* <li>Unit: kbps (kilobits per second).</li>
* <li>Default: 100.</li>
* <li>
* 0 = unconstrained: the video bitrate will drop as needed, even to the
* lowest possible quality, which might make the video completely blurry and
* pixelated.
* </li>
* </ul>
*
* @alias module:core/abstracts.BaseRtpEndpoint#getMinVideoSendBandwidth
*
* @param {module:core/abstracts.BaseRtpEndpoint~getMinVideoSendBandwidthCallback} [callback]
*
* @return {external:Promise}
*/
BaseRtpEndpoint.prototype.getMinVideoSendBandwidth = function(callback){
var transaction = (arguments[0] instanceof Transaction)
? Array.prototype.shift.apply(arguments)
: undefined;
var usePromise = false;
if (callback == undefined) {
usePromise = true;
}
if(!arguments.length) callback = undefined;
callback = (callback || noop).bind(this)
return disguise(this._invoke(transaction, 'getMinVideoSendBandwidth', callback), this)
};
/**
* @callback module:core/abstracts.BaseRtpEndpoint~getMinVideoSendBandwidthCallback
* @param {external:Error} error
* @param {external:Integer} result
*/
/**
* Minimum video bitrate sent to remote peer.
* <p>
* With this parameter you can control the minimum video quality that will be
* sent when reacting to bad network conditions. Setting this parameter to a
* low
* value permits the video quality to drop when the network conditions get
* worse.
* </p>
* <p>
* This parameter provides a way to override the bitrate requested by remote
* REMB
* bandwidth estimations: the bitrate sent will be always equal or greater
* than
* this parameter, even if the remote peer requests even lower bitrates.
* </p>
* <p>
* Note that if you set this parameter too high (trying to avoid bad video
* quality altogether), you would be limiting the adaptation ability of the
* congestion control algorithm, and your stream might be unable to ever
* recover
* from adverse network conditions.
* </p>
* <ul>
* <li>Unit: kbps (kilobits per second).</li>
* <li>Default: 100.</li>
* <li>
* 0 = unconstrained: the video bitrate will drop as needed, even to the
* lowest possible quality, which might make the video completely blurry and
* pixelated.
* </li>
* </ul>
*
* @alias module:core/abstracts.BaseRtpEndpoint#setMinVideoSendBandwidth
*
* @param {external:Integer} minVideoSendBandwidth
* @param {module:core/abstracts.BaseRtpEndpoint~setMinVideoSendBandwidthCallback} [callback]
*
* @return {external:Promise}
*/
BaseRtpEndpoint.prototype.setMinVideoSendBandwidth = function(minVideoSendBandwidth, callback){
var transaction = (arguments[0] instanceof Transaction)
? Array.prototype.shift.apply(arguments)
: undefined;
//
// checkType('int', 'minVideoSendBandwidth', minVideoSendBandwidth, {required: true});
//
var params = {
minVideoSendBandwidth: minVideoSendBandwidth
};
callback = (callback || noop).bind(this)
return disguise(this._invoke(transaction, 'setMinVideoSendBandwidth', params, callback), this)
};
/**
* @callback module:core/abstracts.BaseRtpEndpoint~setMinVideoSendBandwidthCallback
* @param {external:Error} error
*/
/**
* Maximum Transmission Unit (MTU) used for RTP.
* <p>
* This setting affects the maximum size that will be used by RTP payloads.
* You
* can change it from the default, if you think that a different value would
* be
* beneficial for the typical network settings of your application.
* </p>
* <p>
* The default value is 1200 Bytes. This is the same as in <b>libwebrtc</b>
* (from
* webrtc.org), as used by
* <a
* href='https://dxr.mozilla.org/mozilla-central/rev/b5c5ba07d3dbd0d07b66fa42a103f4df2c27d3a2/media/webrtc/trunk/webrtc/media/engine/constants.cc#16'
* >Firefox</a
* >
* or
* <a
* href='https://codesearch.chromium.org/chromium/src/third_party/webrtc/media/engine/constants.cc?l=15&rcl=6dd488b2e55125644263e4837f1abd950d5e410d'
* >Chrome</a
* >
* . You can read more about this value in
* <a
* href='https://groups.google.com/d/topic/discuss-webrtc/gH5ysR3SoZI/discussion'
* >Why RTP max packet size is 1200 in WebRTC?</a
* >
* .
* </p>
* <p>
* <b>WARNING</b>: Change this value ONLY if you really know what you are
* doing
* and you have strong reasons to do so. Do NOT change this parameter just
* because it <i>seems</i> to work better for some reduced scope tests. The
* default value is a consensus chosen by people who have deep knowledge about
* network optimization.
* </p>
* <ul>
* <li>Unit: Bytes.</li>
* <li>Default: 1200.</li>
* </ul>
*
* @alias module:core/abstracts.BaseRtpEndpoint#getMtu
*
* @param {module:core/abstracts.BaseRtpEndpoint~getMtuCallback} [callback]
*
* @return {external:Promise}
*/
BaseRtpEndpoint.prototype.getMtu = function(callback){
var transaction = (arguments[0] instanceof Transaction)
? Array.prototype.shift.apply(arguments)
: undefined;
var usePromise = false;
if (callback == undefined) {
usePromise = true;
}
if(!arguments.length) callback = undefined;
callback = (callback || noop).bind(this)
return disguise(this._invoke(transaction, 'getMtu', callback), this)
};
/**
* @callback module:core/abstracts.BaseRtpEndpoint~getMtuCallback
* @param {external:Error} error
* @param {external:Integer} result
*/
/**
* Maximum Transmission Unit (MTU) used for RTP.
* <p>
* This setting affects the maximum size that will be used by RTP payloads.
* You
* can change it from the default, if you think that a different value would
* be
* beneficial for the typical network settings of your application.
* </p>
* <p>
* The default value is 1200 Bytes. This is the same as in <b>libwebrtc</b>
* (from
* webrtc.org), as used by
* <a
* href='https://dxr.mozilla.org/mozilla-central/rev/b5c5ba07d3dbd0d07b66fa42a103f4df2c27d3a2/media/webrtc/trunk/webrtc/media/engine/constants.cc#16'
* >Firefox</a
* >
* or
* <a
* href='https://codesearch.chromium.org/chromium/src/third_party/webrtc/media/engine/constants.cc?l=15&rcl=6dd488b2e55125644263e4837f1abd950d5e410d'
* >Chrome</a
* >
* . You can read more about this value in
* <a
* href='https://groups.google.com/d/topic/discuss-webrtc/gH5ysR3SoZI/discussion'
* >Why RTP max packet size is 1200 in WebRTC?</a
* >
* .
* </p>
* <p>
* <b>WARNING</b>: Change this value ONLY if you really know what you are
* doing
* and you have strong reasons to do so. Do NOT change this parameter just
* because it <i>seems</i> to work better for some reduced scope tests. The
* default value is a consensus chosen by people who have deep knowledge about
* network optimization.
* </p>
* <ul>
* <li>Unit: Bytes.</li>
* <li>Default: 1200.</li>
* </ul>
*
* @alias module:core/abstracts.BaseRtpEndpoint#setMtu
*
* @param {external:Integer} mtu
* @param {module:core/abstracts.BaseRtpEndpoint~setMtuCallback} [callback]
*
* @return {external:Promise}
*/
BaseRtpEndpoint.prototype.setMtu = function(mtu, callback){
var transaction = (arguments[0] instanceof Transaction)
? Array.prototype.shift.apply(arguments)
: undefined;
//
// checkType('int', 'mtu', mtu, {required: true});
//
var params = {
mtu: mtu
};
callback = (callback || noop).bind(this)
return disguise(this._invoke(transaction, 'setMtu', params, callback), this)
};
/**
* @callback module:core/abstracts.BaseRtpEndpoint~setMtuCallback
* @param {external:Error} error
*/
/**
* Advanced parameters to configure the congestion control algorithm.
*
* @alias module:core/abstracts.BaseRtpEndpoint#getRembParams
*
* @param {module:core/abstracts.BaseRtpEndpoint~getRembParamsCallback} [callback]
*
* @return {external:Promise}
*/
BaseRtpEndpoint.prototype.getRembParams = function(callback){
var transaction = (arguments[0] instanceof Transaction)
? Array.prototype.shift.apply(arguments)
: undefined;
var usePromise = false;
if (callback == undefined) {
usePromise = true;
}
if(!arguments.length) callback = undefined;
callback = (callback || noop).bind(this)
return disguise(this._invoke(transaction, 'getRembParams', callback), this)
};
/**
* @callback module:core/abstracts.BaseRtpEndpoint~getRembParamsCallback
* @param {external:Error} error
* @param {module:core/complexTypes.RembParams} result
*/
/**
* Advanced parameters to configure the congestion control algorithm.
*
* @alias module:core/abstracts.BaseRtpEndpoint#setRembParams
*
* @param {module:core/complexTypes.RembParams} rembParams
* @param {module:core/abstracts.BaseRtpEndpoint~setRembParamsCallback} [callback]
*
* @return {external:Promise}
*/
BaseRtpEndpoint.prototype.setRembParams = function(rembParams, callback){
var transaction = (arguments[0] instanceof Transaction)
? Array.prototype.shift.apply(arguments)
: undefined;
//
// checkType('RembParams', 'rembParams', rembParams, {required: true});
//
var params = {
rembParams: rembParams
};
callback = (callback || noop).bind(this)
return disguise(this._invoke(transaction, 'setRembParams', params, callback), this)
};
/**
* @callback module:core/abstracts.BaseRtpEndpoint~setRembParamsCallback
* @param {external:Error} error
*/
/**
* @alias module:core/abstracts.BaseRtpEndpoint.constructorParams
*/
BaseRtpEndpoint.constructorParams = {
};
/**
* @alias module:core/abstracts.BaseRtpEndpoint.events
*
* @extends module:core/abstracts.SdpEndpoint.events
*/
BaseRtpEndpoint.events = SdpEndpoint.events.concat(['ConnectionStateChanged', 'MediaStateChanged']);
/**
* Checker for {@link module:core/abstracts.BaseRtpEndpoint}
*
* @memberof module:core/abstracts
*
* @param {external:String} key
* @param {module:core/abstracts.BaseRtpEndpoint} value
*/
function checkBaseRtpEndpoint(key, value)
{
if(!(value instanceof BaseRtpEndpoint))
throw ChecktypeError(key, BaseRtpEndpoint, value);
};
module.exports = BaseRtpEndpoint;
BaseRtpEndpoint.check = checkBaseRtpEndpoint;