| Top | |
|
|
|
gstrtpdec acts as a decoder that removes security from SRTP and SRTCP packets (encryption and authentication) and out RTP and RTCP. It receives packet of type 'application/x-srtp' or 'application/x-srtcp' on its sink pad, and outs packets of type 'application/x-rtp' or 'application/x-rtcp' on its source pad.
For each packet received, it checks if the internal SSRC is in the list of streams already in use. If this is not the case, it sends a signal to the user to get the needed parameters to create a new stream : master key, encryption and authentication mecanisms for both RTP and RTCP. If the user can't provide those parameters, the buffer is dropped and a warning is emitted.
This element uses libsrtp library. The encryption and authentication mecanisms available are :
Encryption
AES_ICM 256 bits (maximum security)
AES_ICM 128 bits (default)
NULL
Authentication
HMAC_SHA1 80 bits (default, maximum protection)
HMAC_SHA1 32 bits
NULL
Note that for SRTP protection, authentication is mandatory (non-null) if encryption is used (non-null).
Each packet received is first analysed (checked for valid SSRC) then its buffer is unprotected 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.
When the maximum usage of the master key is reached, a soft-limit signal is sent to the user, and new parameters (master key) are needed in return. 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 the SRTP rollover counter for a given SSRC must be set in the caps "roc" field when the request-key signal is emitted by the decoder. The rollover counters should have been transmitted by a signaling protocol by some other means. If no rollover counter is provided by the user, 0 is used by default.
1 |
gst-launch-1.0 udpsrc port=5004 caps='application/x-srtp, payload=(int)8, ssrc=(uint)1356955624, srtp-key=(buffer)012345678901234567890123456789012345678901234567890123456789, srtp-cipher=(string)aes-128-icm, srtp-auth=(string)hmac-sha1-80, srtcp-cipher=(string)aes-128-icm, srtcp-auth=(string)hmac-sha1-80, roc=(uint)0' ! srtpdec ! rtppcmadepay ! alawdec ! pulsesink |
Receive PCMA SRTP packets through UDP using caps to specify master key and protection.
1 |
gst-launch-1.0 audiotestsrc ! alawenc ! rtppcmapay ! 'application/x-rtp, payload=(int)8, ssrc=(uint)1356955624' ! srtpenc key="012345678901234567890123456789012345678901234567890123456789" ! udpsink port=5004 |
Send PCMA SRTP packets through UDP, nothing how the SSRC is forced so that the receiver will recognize it.
plugin |
srtp |
author |
Gabriel Millaire <millaire.gabriel@collabora.com> |
class |
Filter/Network/SRTP |
name |
rtcp_sink |
direction |
sink |
presence |
always |
details |
application/x-srtcp |
name |
rtp_sink |
direction |
sink |
presence |
always |
details |
application/x-srtp |
name |
rtcp_src |
direction |
source |
presence |
always |
details |
application/x-rtcp |
name |
rtp_src |
direction |
source |
presence |
always |
details |
application/x-rtp |
“replay-window-size” property“replay-window-size” guint
Size of the replay protection window.
Flags: Read / Write
Allowed values: [64,32768]
Default value: 128
“clear-keys” signalvoid user_function (GstSrtpDec *gstsrtpdec, gpointer user_data)
Clear the internal list of streams
gstsrtpdec |
the element on which the signal is emitted |
|
user_data |
user data set when the signal handler was connected. |
Flags: Action
“hard-limit” signalGstCaps* user_function (GstSrtpDec *gstsrtpdec, guint ssrc, gpointer user_data)
Signal emited when the stream with ssrc
has reached the
hard limit of utilisation of it's master encryption key.
User should provide a new key and new RTP and RTCP encryption
ciphers and authentication, and return them wrapped in a
GstCaps. If user could not provide those parameters or signal
is not answered, the buffers of this stream will be dropped.
gstsrtpdec |
the element on which the signal is emitted |
|
ssrc |
The unique SSRC of the stream |
|
user_data |
user data set when the signal handler was connected. |
Flags: Run Last
“request-key” signalGstCaps* user_function (GstSrtpDec *gstsrtpdec, guint ssrc, gpointer user_data)
Signal emited to get the parameters relevant to stream
with ssrc
. User should provide the key and the RTP and
RTCP encryption ciphers and authentication, and return
them wrapped in a GstCaps.
gstsrtpdec |
the element on which the signal is emitted |
|
ssrc |
The unique SSRC of the stream |
|
user_data |
user data set when the signal handler was connected. |
Flags: Run Last
“soft-limit” signalGstCaps* user_function (GstSrtpDec *gstsrtpdec, guint ssrc, gpointer user_data)
Signal emited when the stream with ssrc
has reached the
soft limit of utilisation of it's master encryption key.
User should provide a new key and new RTP and RTCP encryption
ciphers and authentication, and return them wrapped in a
GstCaps.
gstsrtpdec |
the element on which the signal is emitted |
|
ssrc |
The unique SSRC of the stream |
|
user_data |
user data set when the signal handler was connected. |
Flags: Run Last
“remove-key” signalvoid user_function (GstSrtpDec *gstsrtpdec, guint ssrc, gpointer user_data)
Removes keys for a specific SSRC
gstsrtpdec |
the element on which the signal is emitted |
|
ssrc |
The SSRC for which to remove the key. |
|
user_data |
user data set when the signal handler was connected. |
Flags: Action