Properties

Signals

Types and Values

Object Hierarchy

    GObject
    ╰── GInitiallyUnowned
        ╰── GstObject
            ╰── GstElement
                ╰── GstSrtpEnc

Description

gstrtpenc acts as an encoder that adds security to RTP and RTCP packets in the form of encryption and authentication. It outs SRTP and SRTCP.

An application can request multiple RTP and RTCP pads to protect, but every sink pad requested must receive packets from the same source (identical SSRC). If a packet received contains a different SSRC, a warning is emited and the valid SSRC is forced on the packet.

This element uses libsrtp library. When receiving the first packet, the library is initialized with a new stream (based on the SSRC). It uses the default RTP and RTCP encryption and authentication mechanisms, unless the user has set the relevant properties first. It also uses a master key that MUST be set by property (key) at the beginning. The master key must be of a maximum length of 46 characters (14 characters for the salt plus the key). The encryption and authentication mecanisms available are :

Encryption (properties rtp-cipher and rtcp-cipher)

Authentication (properties rtp-auth and rtcp-auth)

Note that for SRTP protection, authentication is mandatory (non-null) if encryption is used (non-null).

When requested to create a sink pad, a linked source pad is created. Each packet received is first analysed (checked for valid SSRC) then its buffer is protected with libsrtp, then pushed on the source pad. If protection failed or the stream could not be created, the buffer is dropped and a warning is emitted. The packets pushed on the source pad are of type 'application/x-srtp' or 'application/x-srtcp'.

When the maximum usage of the master key is reached, a soft-limit signal is sent to the user. The user must then set a new master key by property. If the hard limit is reached, a flag is set and every subsequent packet is dropped, until a new key is set and the stream has been updated.

If a stream is to be shared between multiple clients it is also possible to request the internal SRTP rollover counter for a given SSRC. The rollover counter should be then transmitted and used by the clients to authenticate and decrypt the packets. Failing to do that the clients will start with a rollover counter of 0 which will probably be incorrect if the stream has been transmitted for a while to other clients.

Functions

Types and Values

struct GstSrtpEnc;

Property Details

  “key”                      GstBuffer *

  “random-key”               gboolean

  “rtcp-auth”                GstSrtpAuthType

  “rtcp-cipher”              GstSrtpCipherType

  “rtp-auth”                 GstSrtpAuthType

  “rtp-cipher”               GstSrtpCipherType

  “replay-window-size”       guint

  “allow-repeat-tx”          gboolean

  “stats”                    GstStructure *

Signal Details

void
user_function (GstSrtpEnc *gstsrtpenc,
               gpointer    user_data)

guint
user_function (GstSrtpEnc *gstsrtpenc,
               guint       arg1,
               gpointer    user_data)

See Also

srtpdec