Interface BaseRtpEndpoint

All Superinterfaces:
Endpoint, KurentoObject, MediaElement, MediaObject, SdpEndpoint, SessionEndpoint
All Known Subinterfaces:
RtpEndpoint, WebRtcEndpoint
public interface BaseRtpEndpoint extends SdpEndpoint
Base class for the implementation of RTP-based communication endpoints.

All endpoints that rely on the RTP protocol, like the :rom:cls:`RtpEndpoint` or the :rom:cls:`WebRtcEndpoint`, inherit from this class. This endpoint provides information about the Connection state and the Media state, which can be consulted at any time through the getMediaState() and the getConnectionState() methods. It is also possible subscribe to events fired when these properties change:

  • ConnectionStateChangedEvent: This event is raised when the connection between two peers changes. It can have two values:
    • CONNECTED
    • DISCONNECTED
  • MediaStateChangedEvent: This event provides information about the state of the underlying RTP session. Possible values are:
    • CONNECTED: There is an RTCP packet flow between peers.
    • DISCONNECTED: Either no RTCP packets have been received yet, or the remote peer has ended the RTP session with a BYE message, or at least 5 seconds have elapsed since the last RTCP packet was received.

    The standard definition of RTP (RFC 3550) describes a session as active whenever there is a maintained flow of RTCP control packets, regardless of whether there is actual media flowing through RTP data packets or not. The reasoning behind this is that, at any given moment, a participant of an RTP session might temporarily stop sending RTP data packets, but this wouldn't necessarily mean that the RTP session as a whole is finished; it maybe just means that the participant has some temporary issues but it will soon resume sending data. For this reason, that an RTP session has really finished is something that is considered only by the prolonged absence of RTCP control packets between participants.

    Since RTCP packets do not flow at a constant rate (for instance, minimizing a browser window with a WebRTC's RTCPeerConnection object might affect the sending interval), it is not possible to immediately detect their absence and assume that the RTP session has finished. Instead, there is a guard period of approximately 5 seconds of missing RTCP packets before considering that the underlying RTP session is effectively finished, thus triggering a MediaStateChangedEvent = DISCONNECTED event.

    In other words, there is always a period during which there might be no media flowing, but this event hasn't been fired yet. Nevertheless, this is the most reliable and useful way of knowing what is the long-term, steady state of RTP media exchange.

    The ConnectionStateChangedEvent comes in contrast with more instantaneous events such as MediaElement's MediaFlowInStateChangedEvent and MediaFlowOutStateChangedEvent, which are triggered almost immediately after the RTP data packets stop flowing between RTP session participants. This makes the MediaFlow events a good way to know if participants are suffering from short-term intermittent connectivity issues, but they are not enough to know if the connectivity issues are just spurious network hiccups or are part of a more long-term disconnection problem.

Bitrate management

Bandwidth control for the video component of the media session is done here. Note that the default MaxVideoSendBandwidth is a VERY conservative value, and leads to a low maximum video quality. Most applications will probably want to increase this to higher values such as 2000 kbps (2 Mbps).

  • Recv bandwidth: Used to request a remote sender that its media bitrate is within the requested range.
    • MinVideoRecvBandwidth: Minimum inbound bitrate requested by this endpoint. Signaled to compatible WebRTC and RTP senders, as part of the REMB bandwidth estimation protocol.
      • Unit: kbps (kilobits per second).
      • Default: 0.
      • Note: The actual minimum value is 30 kbps, even if a lower value is set.
    • MaxAudioRecvBandwidth and MaxVideoRecvBandwidth: Maximum inbound bitrate requested by this endpoint. Signaled to compatible WebRTC and RTP senders as part of Kurento's REMB bandwidth estimations, and also as an SDP bitrate attribute (b=AS:{value}, see RFC 8866 Section 5.8. Bandwidth Information) in sendrecv and recvonly SDP Offers.
      • Unit: kbps (kilobits per second).
      • Default: 0 (unlimited).
  • Send bandwidth: Used to control bitrate of the outbound media stream sent to remote peers. It is important to keep in mind that outbound bitrate ultimately depends on network and remote peer capabilities.
    • MinVideoSendBandwidth: Minimum outbound bitrate sent by this endpoint.
      • Unit: kbps (kilobits per second).
      • Default: 100.
      • 0 = unlimited: the video bitrate will drop as needed, even to the lowest possible quality, which could make the video completely blurry and pixelated, but would adapt better to losses caused by network congestion.
    • MaxVideoSendBandwidth: Maximum outbound bitrate sent by this endpoint. Remote peers themselves might also announce a maximum bitrate as part of their REMB bandwidth estimations, and also as an SDP bitrate attribute (b=AS:{value}, see RFC 8866 Section 5.8. Bandwidth Information) in sendrecv and recvonly SDP Offers or Answers. Kurento will always give priority to bitrate limitations specified by remote peers, over internal configurations.
      • Unit: kbps (kilobits per second).
      • Default: 500.
      • 0 = unlimited: the video bitrate will grow until all the available network bandwidth is used by the stream.
        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).
    • RembParams.rembOnConnect: Initial local REMB bandwidth estimation that gets used when a new endpoint is connected. Only useful for connections that are compatible with the REMB bandwidth estimation protocol (such as most WebRTC peers).

All bandwidth control parameters must be set before the SDP negotiation takes place, and can't be modified afterwards.

Take into consideration that setting a too high upper limit for the output bandwidth can be a reason for the network connection to be congested quickly.

  • Method Details

    • getMinVideoRecvBandwidth

      int getMinVideoRecvBandwidth()
      Get Minimum inbound bitrate requested by this endpoint. Signaled to compatible WebRTC and RTP senders, as part of the REMB bandwidth estimation protocol.

      This is used to set a minimum value of local REMB during bandwidth estimation. The REMB estimation will then be sent to remote peers, causing them to send at least the indicated video bitrate. It follows that this parameter will only have effect for remote peers that support the REMB bandwidth estimation protocol (such as e.g. most web browsers compatible with WebRTC).

      • Unit: kbps (kilobits per second).
      • Default: 0.
      • Note: The actual minimum value is 30 kbps, even if a lower value is set.

    • getMinVideoRecvBandwidth

      void getMinVideoRecvBandwidth(Continuation<Integer> cont)
      Get Minimum inbound bitrate requested by this endpoint. Signaled to compatible WebRTC and RTP senders, as part of the REMB bandwidth estimation protocol.

      This is used to set a minimum value of local REMB during bandwidth estimation. The REMB estimation will then be sent to remote peers, causing them to send at least the indicated video bitrate. It follows that this parameter will only have effect for remote peers that support the REMB bandwidth estimation protocol (such as e.g. most web browsers compatible with WebRTC).

      • Unit: kbps (kilobits per second).
      • Default: 0.
      • Note: The actual minimum value is 30 kbps, even if a lower value is set.

    • getMinVideoRecvBandwidth

      TFuture<Integer> getMinVideoRecvBandwidth(Transaction tx)
      Get Minimum inbound bitrate requested by this endpoint. Signaled to compatible WebRTC and RTP senders, as part of the REMB bandwidth estimation protocol.

      This is used to set a minimum value of local REMB during bandwidth estimation. The REMB estimation will then be sent to remote peers, causing them to send at least the indicated video bitrate. It follows that this parameter will only have effect for remote peers that support the REMB bandwidth estimation protocol (such as e.g. most web browsers compatible with WebRTC).

      • Unit: kbps (kilobits per second).
      • Default: 0.
      • Note: The actual minimum value is 30 kbps, even if a lower value is set.

    • setMinVideoRecvBandwidth

      void setMinVideoRecvBandwidth(int minVideoRecvBandwidth)
      Set Minimum inbound bitrate requested by this endpoint. Signaled to compatible WebRTC and RTP senders, as part of the REMB bandwidth estimation protocol.

      This is used to set a minimum value of local REMB during bandwidth estimation. The REMB estimation will then be sent to remote peers, causing them to send at least the indicated video bitrate. It follows that this parameter will only have effect for remote peers that support the REMB bandwidth estimation protocol (such as e.g. most web browsers compatible with WebRTC).

      • Unit: kbps (kilobits per second).
      • Default: 0.
      • Note: The actual minimum value is 30 kbps, even if a lower value is set.

    • setMinVideoRecvBandwidth

      void setMinVideoRecvBandwidth(int minVideoRecvBandwidth, Continuation<Void> cont)
      Set Minimum inbound bitrate requested by this endpoint. Signaled to compatible WebRTC and RTP senders, as part of the REMB bandwidth estimation protocol.

      This is used to set a minimum value of local REMB during bandwidth estimation. The REMB estimation will then be sent to remote peers, causing them to send at least the indicated video bitrate. It follows that this parameter will only have effect for remote peers that support the REMB bandwidth estimation protocol (such as e.g. most web browsers compatible with WebRTC).

      • Unit: kbps (kilobits per second).
      • Default: 0.
      • Note: The actual minimum value is 30 kbps, even if a lower value is set.

    • setMinVideoRecvBandwidth

      void setMinVideoRecvBandwidth(int minVideoRecvBandwidth, Transaction tx)
      Set Minimum inbound bitrate requested by this endpoint. Signaled to compatible WebRTC and RTP senders, as part of the REMB bandwidth estimation protocol.

      This is used to set a minimum value of local REMB during bandwidth estimation. The REMB estimation will then be sent to remote peers, causing them to send at least the indicated video bitrate. It follows that this parameter will only have effect for remote peers that support the REMB bandwidth estimation protocol (such as e.g. most web browsers compatible with WebRTC).

      • Unit: kbps (kilobits per second).
      • Default: 0.
      • Note: The actual minimum value is 30 kbps, even if a lower value is set.

    • getMinVideoSendBandwidth

      int getMinVideoSendBandwidth()
      Get Minimum outbound bitrate sent by this endpoint.

      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.

      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.

      • Unit: kbps (kilobits per second).
      • Default: 100.
      • 0 = unlimited: the video bitrate will drop as needed, even to the lowest possible quality, which could make the video completely blurry and pixelated, but would adapt better to losses caused by network congestion.

    • getMinVideoSendBandwidth

      void getMinVideoSendBandwidth(Continuation<Integer> cont)
      Get Minimum outbound bitrate sent by this endpoint.

      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.

      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.

      • Unit: kbps (kilobits per second).
      • Default: 100.
      • 0 = unlimited: the video bitrate will drop as needed, even to the lowest possible quality, which could make the video completely blurry and pixelated, but would adapt better to losses caused by network congestion.

    • getMinVideoSendBandwidth

      TFuture<Integer> getMinVideoSendBandwidth(Transaction tx)
      Get Minimum outbound bitrate sent by this endpoint.

      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.

      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.

      • Unit: kbps (kilobits per second).
      • Default: 100.
      • 0 = unlimited: the video bitrate will drop as needed, even to the lowest possible quality, which could make the video completely blurry and pixelated, but would adapt better to losses caused by network congestion.

    • setMinVideoSendBandwidth

      void setMinVideoSendBandwidth(int minVideoSendBandwidth)
      Set Minimum outbound bitrate sent by this endpoint.

      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.

      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.

      • Unit: kbps (kilobits per second).
      • Default: 100.
      • 0 = unlimited: the video bitrate will drop as needed, even to the lowest possible quality, which could make the video completely blurry and pixelated, but would adapt better to losses caused by network congestion.

    • setMinVideoSendBandwidth

      void setMinVideoSendBandwidth(int minVideoSendBandwidth, Continuation<Void> cont)
      Set Minimum outbound bitrate sent by this endpoint.

      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.

      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.

      • Unit: kbps (kilobits per second).
      • Default: 100.
      • 0 = unlimited: the video bitrate will drop as needed, even to the lowest possible quality, which could make the video completely blurry and pixelated, but would adapt better to losses caused by network congestion.

    • setMinVideoSendBandwidth

      void setMinVideoSendBandwidth(int minVideoSendBandwidth, Transaction tx)
      Set Minimum outbound bitrate sent by this endpoint.

      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.

      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.

      • Unit: kbps (kilobits per second).
      • Default: 100.
      • 0 = unlimited: the video bitrate will drop as needed, even to the lowest possible quality, which could make the video completely blurry and pixelated, but would adapt better to losses caused by network congestion.

    • getMaxVideoSendBandwidth

      int getMaxVideoSendBandwidth()
      Get Maximum outbound bitrate sent by this endpoint. Remote peers themselves might also announce a maximum bitrate as part of their REMB bandwidth estimations, and also as an SDP bitrate attribute (b=AS:{value}, see RFC 8866 Section 5.8. Bandwidth Information) in sendrecv and recvonly SDP Offers or Answers. Kurento will always give priority to bitrate limitations specified by remote peers, over internal configurations.

      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.

      Note that the default MaxVideoSendBandwidth is a VERY conservative value, and leads to a low maximum video quality. Most applications will probably want to increase this to higher values such as 2000 kbps (2 Mbps).

      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 RembParams.rembOnConnect.

      • Unit: kbps (kilobits per second).
      • Default: 500.
      • 0 = unlimited: the video bitrate will grow until all the available network bandwidth is used by the stream.
        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).

    • getMaxVideoSendBandwidth

      void getMaxVideoSendBandwidth(Continuation<Integer> cont)
      Get Maximum outbound bitrate sent by this endpoint. Remote peers themselves might also announce a maximum bitrate as part of their REMB bandwidth estimations, and also as an SDP bitrate attribute (b=AS:{value}, see RFC 8866 Section 5.8. Bandwidth Information) in sendrecv and recvonly SDP Offers or Answers. Kurento will always give priority to bitrate limitations specified by remote peers, over internal configurations.

      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.

      Note that the default MaxVideoSendBandwidth is a VERY conservative value, and leads to a low maximum video quality. Most applications will probably want to increase this to higher values such as 2000 kbps (2 Mbps).

      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 RembParams.rembOnConnect.

      • Unit: kbps (kilobits per second).
      • Default: 500.
      • 0 = unlimited: the video bitrate will grow until all the available network bandwidth is used by the stream.
        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).

    • getMaxVideoSendBandwidth

      TFuture<Integer> getMaxVideoSendBandwidth(Transaction tx)
      Get Maximum outbound bitrate sent by this endpoint. Remote peers themselves might also announce a maximum bitrate as part of their REMB bandwidth estimations, and also as an SDP bitrate attribute (b=AS:{value}, see RFC 8866 Section 5.8. Bandwidth Information) in sendrecv and recvonly SDP Offers or Answers. Kurento will always give priority to bitrate limitations specified by remote peers, over internal configurations.

      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.

      Note that the default MaxVideoSendBandwidth is a VERY conservative value, and leads to a low maximum video quality. Most applications will probably want to increase this to higher values such as 2000 kbps (2 Mbps).

      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 RembParams.rembOnConnect.

      • Unit: kbps (kilobits per second).
      • Default: 500.
      • 0 = unlimited: the video bitrate will grow until all the available network bandwidth is used by the stream.
        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).

    • setMaxVideoSendBandwidth

      void setMaxVideoSendBandwidth(int maxVideoSendBandwidth)
      Set Maximum outbound bitrate sent by this endpoint. Remote peers themselves might also announce a maximum bitrate as part of their REMB bandwidth estimations, and also as an SDP bitrate attribute (b=AS:{value}, see RFC 8866 Section 5.8. Bandwidth Information) in sendrecv and recvonly SDP Offers or Answers. Kurento will always give priority to bitrate limitations specified by remote peers, over internal configurations.

      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.

      Note that the default MaxVideoSendBandwidth is a VERY conservative value, and leads to a low maximum video quality. Most applications will probably want to increase this to higher values such as 2000 kbps (2 Mbps).

      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 RembParams.rembOnConnect.

      • Unit: kbps (kilobits per second).
      • Default: 500.
      • 0 = unlimited: the video bitrate will grow until all the available network bandwidth is used by the stream.
        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).

    • setMaxVideoSendBandwidth

      void setMaxVideoSendBandwidth(int maxVideoSendBandwidth, Continuation<Void> cont)
      Set Maximum outbound bitrate sent by this endpoint. Remote peers themselves might also announce a maximum bitrate as part of their REMB bandwidth estimations, and also as an SDP bitrate attribute (b=AS:{value}, see RFC 8866 Section 5.8. Bandwidth Information) in sendrecv and recvonly SDP Offers or Answers. Kurento will always give priority to bitrate limitations specified by remote peers, over internal configurations.

      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.

      Note that the default MaxVideoSendBandwidth is a VERY conservative value, and leads to a low maximum video quality. Most applications will probably want to increase this to higher values such as 2000 kbps (2 Mbps).

      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 RembParams.rembOnConnect.

      • Unit: kbps (kilobits per second).
      • Default: 500.
      • 0 = unlimited: the video bitrate will grow until all the available network bandwidth is used by the stream.
        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).

    • setMaxVideoSendBandwidth

      void setMaxVideoSendBandwidth(int maxVideoSendBandwidth, Transaction tx)
      Set Maximum outbound bitrate sent by this endpoint. Remote peers themselves might also announce a maximum bitrate as part of their REMB bandwidth estimations, and also as an SDP bitrate attribute (b=AS:{value}, see RFC 8866 Section 5.8. Bandwidth Information) in sendrecv and recvonly SDP Offers or Answers. Kurento will always give priority to bitrate limitations specified by remote peers, over internal configurations.

      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.

      Note that the default MaxVideoSendBandwidth is a VERY conservative value, and leads to a low maximum video quality. Most applications will probably want to increase this to higher values such as 2000 kbps (2 Mbps).

      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 RembParams.rembOnConnect.

      • Unit: kbps (kilobits per second).
      • Default: 500.
      • 0 = unlimited: the video bitrate will grow until all the available network bandwidth is used by the stream.
        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).

    • getMediaState

      MediaState getMediaState()
      Get Media flow state.
      • CONNECTED: There is an RTCP flow.
      • DISCONNECTED: No RTCP packets have been received for at least 5 sec.

    • getMediaState

      void getMediaState(Continuation<MediaState> cont)
      Get Media flow state.
      • CONNECTED: There is an RTCP flow.
      • DISCONNECTED: No RTCP packets have been received for at least 5 sec.

    • getMediaState

      TFuture<MediaState> getMediaState(Transaction tx)
      Get Media flow state.
      • CONNECTED: There is an RTCP flow.
      • DISCONNECTED: No RTCP packets have been received for at least 5 sec.

    • getConnectionState

      ConnectionState getConnectionState()
      Get Connection state.
      • CONNECTED
      • DISCONNECTED

    • getConnectionState

      void getConnectionState(Continuation<ConnectionState> cont)
      Get Connection state.
      • CONNECTED
      • DISCONNECTED

    • getConnectionState

      TFuture<ConnectionState> getConnectionState(Transaction tx)
      Get Connection state.
      • CONNECTED
      • DISCONNECTED

    • getMtu

      int getMtu()
      Get Maximum Transmission Unit (MTU) used for RTP.

      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.

      The default value is 1200 Bytes. This is the same as in libwebrtc (from webrtc.org), as used by Firefox or Chrome . You can read more about this value in Why RTP max packet size is 1200 in WebRTC? .

      WARNING: 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 seems to work better for some reduced scope tests. The default value is a consensus chosen by people who have deep knowledge about network optimization.

      • Unit: Bytes.
      • Default: 1200.

    • getMtu

      void getMtu(Continuation<Integer> cont)
      Get Maximum Transmission Unit (MTU) used for RTP.

      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.

      The default value is 1200 Bytes. This is the same as in libwebrtc (from webrtc.org), as used by Firefox or Chrome . You can read more about this value in Why RTP max packet size is 1200 in WebRTC? .

      WARNING: 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 seems to work better for some reduced scope tests. The default value is a consensus chosen by people who have deep knowledge about network optimization.

      • Unit: Bytes.
      • Default: 1200.

    • getMtu

      TFuture<Integer> getMtu(Transaction tx)
      Get Maximum Transmission Unit (MTU) used for RTP.

      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.

      The default value is 1200 Bytes. This is the same as in libwebrtc (from webrtc.org), as used by Firefox or Chrome . You can read more about this value in Why RTP max packet size is 1200 in WebRTC? .

      WARNING: 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 seems to work better for some reduced scope tests. The default value is a consensus chosen by people who have deep knowledge about network optimization.

      • Unit: Bytes.
      • Default: 1200.

    • setMtu

      void setMtu(int mtu)
      Set Maximum Transmission Unit (MTU) used for RTP.

      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.

      The default value is 1200 Bytes. This is the same as in libwebrtc (from webrtc.org), as used by Firefox or Chrome . You can read more about this value in Why RTP max packet size is 1200 in WebRTC? .

      WARNING: 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 seems to work better for some reduced scope tests. The default value is a consensus chosen by people who have deep knowledge about network optimization.

      • Unit: Bytes.
      • Default: 1200.

    • setMtu

      void setMtu(int mtu, Continuation<Void> cont)
      Set Maximum Transmission Unit (MTU) used for RTP.

      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.

      The default value is 1200 Bytes. This is the same as in libwebrtc (from webrtc.org), as used by Firefox or Chrome . You can read more about this value in Why RTP max packet size is 1200 in WebRTC? .

      WARNING: 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 seems to work better for some reduced scope tests. The default value is a consensus chosen by people who have deep knowledge about network optimization.

      • Unit: Bytes.
      • Default: 1200.

    • setMtu

      void setMtu(int mtu, Transaction tx)
      Set Maximum Transmission Unit (MTU) used for RTP.

      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.

      The default value is 1200 Bytes. This is the same as in libwebrtc (from webrtc.org), as used by Firefox or Chrome . You can read more about this value in Why RTP max packet size is 1200 in WebRTC? .

      WARNING: 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 seems to work better for some reduced scope tests. The default value is a consensus chosen by people who have deep knowledge about network optimization.

      • Unit: Bytes.
      • Default: 1200.

    • getRembParams

      RembParams getRembParams()
      Get Advanced parameters to configure the congestion control algorithm.

    • getRembParams

      void getRembParams(Continuation<RembParams> cont)
      Get Advanced parameters to configure the congestion control algorithm.

    • getRembParams

      TFuture<RembParams> getRembParams(Transaction tx)
      Get Advanced parameters to configure the congestion control algorithm.

    • setRembParams

      void setRembParams(RembParams rembParams)
      Set Advanced parameters to configure the congestion control algorithm.

    • setRembParams

      void setRembParams(RembParams rembParams, Continuation<Void> cont)
      Set Advanced parameters to configure the congestion control algorithm.

    • setRembParams

      void setRembParams(RembParams rembParams, Transaction tx)
      Set Advanced parameters to configure the congestion control algorithm.

    • requestKeyframe

      void requestKeyframe()
      Force sending a new keyframe request to the upstream elements in the Kurento Pipeline, towards the associated producer. Only valid for video consumers.

    • requestKeyframe

      void requestKeyframe(Continuation<Void> cont)
      Asynchronous version of requestKeyframe: Continuation.onSuccess(F) is called when the action is done. If an error occurs, Continuation.onError(Throwable) is called.
      See Also:

    • requestKeyframe

      void requestKeyframe(Transaction tx)
      Force sending a new keyframe request to the upstream elements in the Kurento Pipeline, towards the associated producer. Only valid for video consumers.

    • addMediaStateChangedListener

      ListenerSubscription addMediaStateChangedListener(EventListener<MediaStateChangedEvent> listener)
      Add a EventListener for event MediaStateChangedEvent. Synchronous call.
      Parameters:
      listener - Listener to be called on MediaStateChangedEvent
      Returns:
      ListenerSubscription for the given Listener

    • addMediaStateChangedListener

      void addMediaStateChangedListener(EventListener<MediaStateChangedEvent> listener, Continuation<ListenerSubscription> cont)
      Add a EventListener for event MediaStateChangedEvent. Asynchronous call. Calls Continuation<ListenerSubscription> when it has been added.
      Parameters:
      listener - Listener to be called on MediaStateChangedEvent
      cont - Continuation to be called when the listener is registered

    • removeMediaStateChangedListener

      void removeMediaStateChangedListener(ListenerSubscription listenerSubscription)
      Remove a ListenerSubscription for event MediaStateChangedEvent. Synchronous call.
      Parameters:
      listenerSubscription - Listener subscription to be removed

    • removeMediaStateChangedListener

      void removeMediaStateChangedListener(ListenerSubscription listenerSubscription, Continuation<Void> cont)
      Remove a ListenerSubscription for event MediaStateChangedEvent. Asynchronous call. Calls Continuation<Void> when it has been removed.
      Parameters:
      listenerSubscription - Listener subscription to be removed
      cont - Continuation to be called when the listener is removed

    • addConnectionStateChangedListener

      ListenerSubscription addConnectionStateChangedListener(EventListener<ConnectionStateChangedEvent> listener)
      Add a EventListener for event ConnectionStateChangedEvent. Synchronous call.
      Parameters:
      listener - Listener to be called on ConnectionStateChangedEvent
      Returns:
      ListenerSubscription for the given Listener

    • addConnectionStateChangedListener

      void addConnectionStateChangedListener(EventListener<ConnectionStateChangedEvent> listener, Continuation<ListenerSubscription> cont)
      Add a EventListener for event ConnectionStateChangedEvent. Asynchronous call. Calls Continuation<ListenerSubscription> when it has been added.
      Parameters:
      listener - Listener to be called on ConnectionStateChangedEvent
      cont - Continuation to be called when the listener is registered

    • removeConnectionStateChangedListener

      void removeConnectionStateChangedListener(ListenerSubscription listenerSubscription)
      Remove a ListenerSubscription for event ConnectionStateChangedEvent. Synchronous call.
      Parameters:
      listenerSubscription - Listener subscription to be removed

    • removeConnectionStateChangedListener

      void removeConnectionStateChangedListener(ListenerSubscription listenerSubscription, Continuation<Void> cont)
      Remove a ListenerSubscription for event ConnectionStateChangedEvent. Asynchronous call. Calls Continuation<Void> when it has been removed.
      Parameters:
      listenerSubscription - Listener subscription to be removed
      cont - Continuation to be called when the listener is removed