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rtpbin

rtpbin

Properties

gboolean do-lost Read / Write
guint latency Read / Write
GstStructure * sdes Read / Write
gboolean ignore-pt Read / Write
gboolean autoremove Read / Write
RTPJitterBufferMode buffer-mode Read / Write
gboolean ntp-sync Read / Write
gboolean use-pipeline-clock Read / Write
GstRTCPSync rtcp-sync Read / Write
guint rtcp-sync-interval Read / Write
gboolean drop-on-latency Read / Write
gboolean do-sync-event Read / Write
gboolean do-retransmission Read / Write
GstRTPProfile rtp-profile Read / Write
GstRtpNtpTimeSource ntp-time-source Read / Write
guint max-dropout-time Read / Write
guint max-misorder-time Read / Write
gint max-rtcp-rtp-time-diff Read / Write
gboolean rtcp-sync-send-time Read / Write
gboolean rfc7273-sync Read / Write
guint max-streams Read / Write

Types and Values

struct GstRtpBin

Object Hierarchy

    GObject
    ╰── GInitiallyUnowned
        ╰── GstObject
            ╰── GstElement
                ╰── GstBin
                    ╰── GstRtpBin

Implemented Interfaces

GstRtpBin implements GstChildProxy.

Description

RTP bin combines the functions of GstRtpSession, GstRtpSsrcDemux, GstRtpJitterBuffer and GstRtpPtDemux in one element. It allows for multiple RTP sessions that will be synchronized together using RTCP SR packets.

GstRtpBin is configured with a number of request pads that define the functionality that is activated, similar to the GstRtpSession element.

To use GstRtpBin as an RTP receiver, request a recv_rtp_sink_%u pad. The session number must be specified in the pad name. Data received on the recv_rtp_sink_%u pad will be processed in the GstRtpSession manager and after being validated forwarded on GstRtpSsrcDemux element. Each RTP stream is demuxed based on the SSRC and send to a GstRtpJitterBuffer. After the packets are released from the jitterbuffer, they will be forwarded to a GstRtpPtDemux element. The GstRtpPtDemux element will demux the packets based on the payload type and will create a unique pad recv_rtp_src_%u_%u_%u on rtpbin with the session number, SSRC and payload type respectively as the pad name.

To also use GstRtpBin as an RTCP receiver, request a recv_rtcp_sink_%u pad. The session number must be specified in the pad name.

If you want the session manager to generate and send RTCP packets, request the send_rtcp_src_%u pad with the session number in the pad name. Packet pushed on this pad contain SR/RR RTCP reports that should be sent to all participants in the session.

To use GstRtpBin as a sender, request a send_rtp_sink_%u pad, which will automatically create a send_rtp_src_%u pad. If the session number is not provided, the pad from the lowest available session will be returned. The session manager will modify the SSRC in the RTP packets to its own SSRC and wil forward the packets on the send_rtp_src_%u pad after updating its internal state.

GstRtpBin can also demultiplex incoming bundled streams. The first GstRtpSession will have a GstRtpSsrcDemux element splitting the streams based on their SSRC and potentially dispatched to a different GstRtpSession. Because retransmission SSRCs need to be merged with the corresponding media stream the “on-bundled-ssrc” signal is emitted so that the application can find out to which session the SSRC belongs.

The session manager needs the clock-rate of the payload types it is handling and will signal the “request-pt-map” signal when it needs such a mapping. One can clear the cached values with the “clear-pt-map” signal.

Access to the internal statistics of rtpbin is provided with the get-internal-session property. This action signal gives access to the RTPSession object which further provides action signals to retrieve the internal source and other sources.

GstRtpBin also has signals (“request-rtp-encoder”, “request-rtp-decoder”, “request-rtcp-encoder” and “request-rtp-decoder”) to dynamically request for RTP and RTCP encoders and decoders in order to support SRTP. The encoders must provide the pads rtp_sink_%u and rtp_src_%u for RTP and rtcp_sink_%u and rtcp_src_%u for RTCP. The session number will be used in the pad name. The decoders must provide rtp_sink and rtp_src for RTP and rtcp_sink and rtcp_src for RTCP. The decoders will be placed before the GstRtpSession element, thus they must support SSRC demuxing internally.

GstRtpBin has signals (“request-aux-sender” and “request-aux-receiver” to dynamically request an element that can be used to create or merge additional RTP streams. AUX elements are needed to implement FEC or retransmission (such as RFC 4588). An AUX sender must have one sink_%u pad that matches the sessionid in the signal and it should have 1 or more src_%u pads. For each src_%\u pad, a session will be made (if needed) and the pad will be linked to the session send_rtp_sink pad. Each session will then expose its source pad as send_rtp_src_%u on GstRtpBin. An AUX receiver has 1 src_%u pad that much match the sessionid in the signal and 1 or more sink_%u pads. A session will be made for each sink_%u pad when the corresponding recv_rtp_sink_%u pad is requested on GstRtpBin.

Example pipelines

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gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
    rtpbin ! rtptheoradepay ! theoradec ! xvimagesink
Receive RTP data from port 5000 and send to the session 0 in rtpbin.
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gst-launch-1.0 rtpbin name=rtpbin \
        v4l2src ! videoconvert ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
                  rtpbin.send_rtp_src_0 ! udpsink port=5000                            \
                  rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false    \
                  udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0                           \
        audiotestsrc ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1                   \
                  rtpbin.send_rtp_src_1 ! udpsink port=5002                            \
                  rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false    \
                  udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
Encode and payload H263 video captured from a v4l2src. Encode and payload AMR audio generated from audiotestsrc. The video is sent to session 0 in rtpbin and the audio is sent to session 1. Video packets are sent on UDP port 5000 and audio packets on port 5002. The video RTCP packets for session 0 are sent on port 5001 and the audio RTCP packets for session 0 are sent on port 5003. RTCP packets for session 0 are received on port 5005 and RTCP for session 1 is received on port 5007. Since RTCP packets from the sender should be sent as soon as possible and do not participate in preroll, sync=false and async=false is configured on udpsink
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gst-launch-1.0 -v rtpbin name=rtpbin                                          \
    udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" \
            port=5000 ! rtpbin.recv_rtp_sink_0                                \
        rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink                    \
     udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0                               \
     rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false        \
    udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1" \
            port=5002 ! rtpbin.recv_rtp_sink_1                                \
        rtpbin. ! rtpamrdepay ! amrnbdec ! alsasink                           \
     udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1                               \
     rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false
Receive H263 on port 5000, send it through rtpbin in session 0, depayload, decode and display the video. Receive AMR on port 5002, send it through rtpbin in session 1, depayload, decode and play the audio. Receive server RTCP packets for session 0 on port 5001 and RTCP packets for session 1 on port 5003. These packets will be used for session management and synchronisation. Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1 on port 5007.

Synopsis

Element Information

plugin

rtpmanager

author

Wim Taymans <wim.taymans@gmail.com>

class

Filter/Network/RTP

Element Pads

name

recv_rtcp_sink_%u

direction

sink

presence

request

details

application/x-rtcp

application/x-srtcp

name

recv_rtp_sink_%u

direction

sink

presence

request

details

application/x-rtp

application/x-srtp

name

send_rtp_sink_%u

direction

sink

presence

request

details

application/x-rtp

name

recv_rtp_src_%u_%u_%u

direction

source

presence

sometimes

details

application/x-rtp

name

send_rtp_src_%u

direction

source

presence

sometimes

details

application/x-rtp

application/x-srtp

name

send_rtcp_src_%u

direction

source

presence

request

details

application/x-rtcp

application/x-srtcp

Functions

Types and Values

struct GstRtpBin

struct GstRtpBin;

Property Details

The “do-lost” property

  “do-lost”                  gboolean

Send an event downstream when a packet is lost.

Flags: Read / Write

Default value: FALSE


The “latency” property

  “latency”                  guint

Default amount of ms to buffer in the jitterbuffers.

Flags: Read / Write

Default value: 200


The “sdes” property

  “sdes”                     GstStructure *

The SDES items of this session.

Flags: Read / Write


The “ignore-pt” property

  “ignore-pt”                gboolean

Do not demultiplex based on PT values.

Flags: Read / Write

Default value: FALSE


The “autoremove” property

  “autoremove”               gboolean

Automatically remove timed out sources.

Flags: Read / Write

Default value: FALSE


The “buffer-mode” property

  “buffer-mode”              RTPJitterBufferMode

Control the buffering and timestamping mode used by the jitterbuffer.

Flags: Read / Write

Default value: Slave receiver to sender clock


The “ntp-sync” property

  “ntp-sync”                 gboolean

Set the NTP time from the sender reports as the running-time on the buffers. When both the sender and receiver have sychronized running-time, i.e. when the clock and base-time is shared between the receivers and the and the senders, this option can be used to synchronize receivers on multiple machines.

Flags: Read / Write

Default value: FALSE


The “use-pipeline-clock” property

  “use-pipeline-clock”       gboolean

Use the pipeline running-time to set the NTP time in the RTCP SR messages (DEPRECATED: Use ntp-time-source property).

Flags: Read / Write

Default value: FALSE


The “rtcp-sync” property

  “rtcp-sync”                GstRTCPSync

If not synchronizing (directly) to the NTP clock, determines how to sync the various streams.

Flags: Read / Write

Default value: always


The “rtcp-sync-interval” property

  “rtcp-sync-interval”       guint

Determines how often to sync streams using RTCP data.

Flags: Read / Write

Default value: 0


The “drop-on-latency” property

  “drop-on-latency”          gboolean

Tells the jitterbuffer to never exceed the given latency in size.

Flags: Read / Write

Default value: FALSE


The “do-sync-event” property

  “do-sync-event”            gboolean

Send event downstream when a stream is synchronized to the sender.

Flags: Read / Write

Default value: FALSE


The “do-retransmission” property

  “do-retransmission”        gboolean

Enables RTP retransmission on all streams. To control retransmission on a per-SSRC basis, connect to the “new-jitterbuffer” signal and set the “do-retransmission” property on the GstRtpJitterBuffer object instead.

Flags: Read / Write

Default value: FALSE


The “rtp-profile” property

  “rtp-profile”              GstRTPProfile

Sets the default RTP profile of newly created RTP sessions. The profile can be changed afterwards on a per-session basis.

Flags: Read / Write

Default value: GST_RTP_PROFILE_AVP


The “ntp-time-source” property

  “ntp-time-source”          GstRtpNtpTimeSource

NTP time source for RTCP packets.

Flags: Read / Write

Default value: NTP time based on realtime clock


The “max-dropout-time” property

  “max-dropout-time”         guint

The maximum time (milliseconds) of missing packets tolerated.

Flags: Read / Write

Default value: 60000


The “max-misorder-time” property

  “max-misorder-time”        guint

The maximum time (milliseconds) of misordered packets tolerated.

Flags: Read / Write

Default value: 2000


The “max-rtcp-rtp-time-diff” property

  “max-rtcp-rtp-time-diff”   gint

Maximum amount of time in ms that the RTP time in RTCP SRs is allowed to be ahead (-1 disabled).

Flags: Read / Write

Allowed values: >= G_MAXULONG

Default value: 1000


The “rtcp-sync-send-time” property

  “rtcp-sync-send-time”      gboolean

Use send time or capture time for RTCP sync (TRUE = send time, FALSE = capture time).

Flags: Read / Write

Default value: TRUE


The “rfc7273-sync” property

  “rfc7273-sync”             gboolean

Synchronize received streams to the RFC7273 clock (requires clock and offset to be provided).

Flags: Read / Write

Default value: FALSE


The “max-streams” property

  “max-streams”              guint

The maximum number of streams to create for one session.

Flags: Read / Write

Default value: 4294967295

Signal Details

The “clear-pt-map” signal

void
user_function (GstRtpBin *rtpbin,
               gpointer   user_data)

Clear all previously cached pt-mapping obtained with “request-pt-map”.

Parameters

rtpbin

the object which received the signal

 

user_data

user data set when the signal handler was connected.

 

Flags: Action


The “get-internal-session” signal

RTPSession*
user_function (GstRtpBin *rtpbin,
               guint      id,
               gpointer   user_data)

Request the internal RTPSession object as GObject in session id .

Parameters

rtpbin

the object which received the signal

 

id

the session id

 

user_data

user data set when the signal handler was connected.

 

Flags: Action


The “on-bye-ssrc” signal

void
user_function (GstRtpBin *rtpbin,
               guint      session,
               guint      ssrc,
               gpointer   user_data)

Notify of an SSRC that became inactive because of a BYE packet.

Parameters

rtpbin

the object which received the signal

 

session

the session

 

ssrc

the SSRC

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last


The “on-bye-timeout” signal

void
user_function (GstRtpBin *rtpbin,
               guint      session,
               guint      ssrc,
               gpointer   user_data)

Notify of an SSRC that has timed out because of BYE

Parameters

rtpbin

the object which received the signal

 

session

the session

 

ssrc

the SSRC

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last


The “on-new-ssrc” signal

void
user_function (GstRtpBin *rtpbin,
               guint      session,
               guint      ssrc,
               gpointer   user_data)

Notify of a new SSRC that entered session .

Parameters

rtpbin

the object which received the signal

 

session

the session

 

ssrc

the SSRC

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last


The “on-npt-stop” signal

void
user_function (GstRtpBin *rtpbin,
               guint      session,
               guint      ssrc,
               gpointer   user_data)

Notify that SSRC sender has sent data up to the configured NPT stop time.

Parameters

rtpbin

the object which received the signal

 

session

the session

 

ssrc

the SSRC

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last


The “on-sender-timeout” signal

void
user_function (GstRtpBin *rtpbin,
               guint      session,
               guint      ssrc,
               gpointer   user_data)

Notify of a sender SSRC that has timed out and became a receiver

Parameters

rtpbin

the object which received the signal

 

session

the session

 

ssrc

the SSRC

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last


The “on-ssrc-active” signal

void
user_function (GstRtpBin *rtpbin,
               guint      session,
               guint      ssrc,
               gpointer   user_data)

Notify of a SSRC that is active, i.e., sending RTCP.

Parameters

rtpbin

the object which received the signal

 

session

the session

 

ssrc

the SSRC

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last


The “on-ssrc-collision” signal

void
user_function (GstRtpBin *rtpbin,
               guint      session,
               guint      ssrc,
               gpointer   user_data)

Notify when we have an SSRC collision

Parameters

rtpbin

the object which received the signal

 

session

the session

 

ssrc

the SSRC

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last


The “on-ssrc-sdes” signal

void
user_function (GstRtpBin *rtpbin,
               guint      session,
               guint      ssrc,
               gpointer   user_data)

Notify of a SSRC that is active, i.e., sending RTCP.

Parameters

rtpbin

the object which received the signal

 

session

the session

 

ssrc

the SSRC

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last


The “on-ssrc-validated” signal

void
user_function (GstRtpBin *rtpbin,
               guint      session,
               guint      ssrc,
               gpointer   user_data)

Notify of a new SSRC that became validated.

Parameters

rtpbin

the object which received the signal

 

session

the session

 

ssrc

the SSRC

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last


The “on-timeout” signal

void
user_function (GstRtpBin *rtpbin,
               guint      session,
               guint      ssrc,
               gpointer   user_data)

Notify of an SSRC that has timed out

Parameters

rtpbin

the object which received the signal

 

session

the session

 

ssrc

the SSRC

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last


The “request-pt-map” signal

GstCaps*
user_function (GstRtpBin *rtpbin,
               guint      session,
               guint      pt,
               gpointer   user_data)

Request the payload type as GstCaps for pt in session .

Parameters

rtpbin

the object which received the signal

 

session

the session

 

pt

the pt

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last


The “reset-sync” signal

void
user_function (GstRtpBin *rtpbin,
               gpointer   user_data)

Reset all currently configured lip-sync parameters and require new SR packets for all streams before lip-sync is attempted again.

Parameters

rtpbin

the object which received the signal

 

user_data

user data set when the signal handler was connected.

 

Flags: Action


The “payload-type-change” signal

void
user_function (GstRtpBin *rtpbin,
               guint      session,
               guint      pt,
               gpointer   user_data)

Signal that the current payload type changed to pt in session .

Parameters

rtpbin

the object which received the signal

 

session

the session

 

pt

the pt

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last


The “new-jitterbuffer” signal

void
user_function (GstRtpBin  *rtpbin,
               GstElement *jitterbuffer,
               guint       session,
               guint       ssrc,
               gpointer    user_data)

Notify that a new jitterbuffer was created for session and ssrc . This signal can, for example, be used to configure jitterbuffer .

Parameters

rtpbin

the object which received the signal

 

jitterbuffer

the new jitterbuffer

 

session

the session

 

ssrc

the SSRC

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last

Since: 1.4


The “request-aux-receiver” signal

GstElement*
user_function (GstRtpBin *rtpbin,
               guint      session,
               gpointer   user_data)

Request an AUX receiver element for the given session . The AUX element will be added to the bin.

If no handler is connected, no AUX element will be used.

Parameters

rtpbin

the object which received the signal

 

session

the session

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last

Since: 1.4


The “request-aux-sender” signal

GstElement*
user_function (GstRtpBin *rtpbin,
               guint      session,
               gpointer   user_data)

Request an AUX sender element for the given session . The AUX element will be added to the bin.

If no handler is connected, no AUX element will be used.

Parameters

rtpbin

the object which received the signal

 

session

the session

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last

Since: 1.4


The “request-rtcp-decoder” signal

GstElement*
user_function (GstRtpBin *rtpbin,
               guint      session,
               gpointer   user_data)

Request an RTCP decoder element for the given session . The decoder element will be added to the bin if not previously added.

If no handler is connected, no encoder will be used.

Parameters

rtpbin

the object which received the signal

 

session

the session

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last

Since: 1.4


The “request-rtcp-encoder” signal

GstElement*
user_function (GstRtpBin *rtpbin,
               guint      session,
               gpointer   user_data)

Request an RTCP encoder element for the given session . The encoder element will be added to the bin if not previously added.

If no handler is connected, no encoder will be used.

Parameters

rtpbin

the object which received the signal

 

session

the session

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last

Since: 1.4


The “request-rtp-decoder” signal

GstElement*
user_function (GstRtpBin *rtpbin,
               guint      session,
               gpointer   user_data)

Request an RTP decoder element for the given session . The decoder element will be added to the bin if not previously added.

If no handler is connected, no encoder will be used.

Parameters

rtpbin

the object which received the signal

 

session

the session

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last

Since: 1.4


The “request-rtp-encoder” signal

GstElement*
user_function (GstRtpBin *rtpbin,
               guint      session,
               gpointer   user_data)

Request an RTP encoder element for the given session . The encoder element will be added to the bin if not previously added.

If no handler is connected, no encoder will be used.

Parameters

rtpbin

the object which received the signal

 

session

the session

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last

Since: 1.4


The “on-new-sender-ssrc” signal

void
user_function (GstRtpBin *rtpbin,
               guint      session,
               guint      ssrc,
               gpointer   user_data)

Notify of a new sender SSRC that entered session .

Parameters

rtpbin

the object which received the signal

 

session

the session

 

ssrc

the sender SSRC

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last

Since: 1.8


The “on-sender-ssrc-active” signal

void
user_function (GstRtpBin *rtpbin,
               guint      session,
               guint      ssrc,
               gpointer   user_data)

Notify of a sender SSRC that is active, i.e., sending RTCP.

Parameters

rtpbin

the object which received the signal

 

session

the session

 

ssrc

the sender SSRC

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last

Since: 1.8


The “get-session” signal

GstElement*
user_function (GstRtpBin *rtpbin,
               guint      id,
               gpointer   user_data)

Request the related GstRtpSession as GstElement related with session id .

Parameters

rtpbin

the object which received the signal

 

id

the session id

 

user_data

user data set when the signal handler was connected.

 

Flags: Action

Since: 1.8


The “on-bundled-ssrc” signal

guint
user_function (GstRtpBin *rtpbin,
               guint      ssrc,
               gpointer   user_data)

Notify of a new incoming bundled SSRC. If no handler is connected to the signal then the GstRtpSession created for the recv_rtp_sink_%u request pad will be managing this new SSRC. However if there is a handler connected then the application can decided to dispatch this new stream to another session by providing its ID as return value of the handler. This can be particularly useful to keep retransmission SSRCs grouped with the session for which they handle retransmission.

Parameters

rtpbin

the object which received the signal

 

ssrc

the bundled SSRC

 

user_data

user data set when the signal handler was connected.

 

Flags: Run Last

Since: 1.12

See Also

rtpjitterbuffer, rtpsession, rtpptdemux, rtpssrcdemux

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