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sip: media-pipeline: Introduce SRTP elements

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The rtpbin will request GstSrtpDec and GstSrtpEnc elements using the
"request-{rtp,rtcp}-{de,en}coder" family of signals.

The newly added boolean use_srtp controls whether the srtp elements are
returned in the signal handler and thus decides if SRTP is used or not.
This commit is contained in:
Evangelos Ribeiro Tzaras 2022-04-07 11:42:53 +02:00
parent 5d0ae4a6fa
commit bfda8f6a3e
1 changed files with 127 additions and 3 deletions
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130
plugins/sip/calls-sip-media-pipeline.c
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@ -146,6 +146,17 @@ struct _CallsSipMediaPipeline {
GstElement *depayloader;
GstElement *decoder;
/* SRTP */
gboolean use_srtp;
GstElement *srtpenc;
GstElement *srtpdec;
gulong request_rtpbin_rtp_decoder_id;
gulong request_rtpbin_rtp_encoder_id;
gulong request_rtpbin_rtcp_encoder_id;
gulong request_rtpbin_rtcp_decoder_id;
/* Gstreamer busses */
GstBus *bus;
guint bus_watch_id;
@ -362,6 +373,51 @@ on_bus_message (GstBus *bus,
}
/* SRTP setup */
static GstCaps *
on_srtpdec_request_key (GstElement *srtpdec,
guint ssrc,
gpointer user_data)
{
/* TODO get key */
return gst_caps_new_simple ("application/x-srtp",
"srtp-cipher", G_TYPE_STRING, "null",
"srtcp-cipher", G_TYPE_STRING, "null",
"srtp-auth", G_TYPE_STRING, "null",
"srtcp-auth", G_TYPE_STRING, "null",
NULL);
}
static GstElement *
on_rtpbin_request_decoder (GstElement *rtpbin,
guint session_id,
gpointer user_data)
{
CallsSipMediaPipeline *self = CALLS_SIP_MEDIA_PIPELINE (user_data);
if (!self->use_srtp)
return NULL;
return gst_object_ref (self->srtpdec);
}
static GstElement *
on_rtpbin_request_encoder (GstElement *rtpbin,
guint session_id,
gpointer user_data)
{
CallsSipMediaPipeline *self = CALLS_SIP_MEDIA_PIPELINE (user_data);
if (!self->use_srtp)
return NULL;
return gst_object_ref (self->srtpenc);
}
/* Pipeline setup */
static gboolean
@ -416,6 +472,7 @@ static gboolean
pipeline_init (CallsSipMediaPipeline *self,
GError **error)
{
GstPad *tmppad;
const char *env_var;
g_assert (CALLS_SIP_MEDIA_PIPELINE (self));
@ -478,6 +535,56 @@ pipeline_init (CallsSipMediaPipeline *self,
/* rtpbin */
MAKE_ELEMENT (rtpbin, "rtpbin", "rtpbin");
/* srtp elements */
MAKE_ELEMENT (srtpdec, "srtpdec", "srtpdec");
g_signal_connect (self->srtpdec,
"request-key",
G_CALLBACK (on_srtpdec_request_key),
self);
MAKE_ELEMENT (srtpenc, "srtpenc", "srtpenc");
g_object_set (self->srtpenc,
"rtp-cipher", 0, "rtp-auth", 0, "rtcp-cipher", 0, "rtcp-auth", 0, NULL);
#if GST_CHECK_VERSION (1, 20, 0)
tmppad = gst_element_request_pad_simple (self->srtpenc, "rtp_sink_0");
#else
tmppad = gst_element_get_request_pad (self->srtpenc, "rtp_sink_0");
#endif
gst_object_unref (tmppad);
#if GST_CHECK_VERSION (1, 20, 0)
tmppad = gst_element_request_pad_simple (self->srtpenc, "rtcp_sink_0");
#else
tmppad = gst_element_get_request_pad (self->srtpenc, "rtcp_sink_0");
#endif
gst_object_unref (tmppad);
self->request_rtpbin_rtp_encoder_id =
g_signal_connect (self->rtpbin,
"request-rtp-encoder",
G_CALLBACK (on_rtpbin_request_encoder),
self);
self->request_rtpbin_rtp_decoder_id =
g_signal_connect (self->rtpbin,
"request-rtp-decoder",
G_CALLBACK (on_rtpbin_request_decoder),
self);
self->request_rtpbin_rtcp_encoder_id =
g_signal_connect (self->rtpbin,
"request-rtcp-encoder",
G_CALLBACK (on_rtpbin_request_encoder),
self);
self->request_rtpbin_rtcp_decoder_id =
g_signal_connect (self->rtpbin,
"request-rtcp-decoder",
G_CALLBACK (on_rtpbin_request_decoder),
self);
/* UDP sources and sinks for RTP and RTCP */
MAKE_ELEMENT (rtp_src, "udpsrc", "rtp-udp-src");
MAKE_ELEMENT (rtp_sink, "udpsink", "rtp-udp-sink");
@ -535,6 +642,7 @@ pipeline_link_elements (CallsSipMediaPipeline *self,
{
g_autoptr (GstPad) srcpad = NULL;
g_autoptr (GstPad) sinkpad = NULL;
GstPadLinkReturn ret;
g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
@ -562,7 +670,8 @@ pipeline_link_elements (CallsSipMediaPipeline *self,
#else
sinkpad = gst_element_get_request_pad (self->rtpbin, "recv_rtp_sink_0");
#endif
if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
ret = gst_pad_link (srcpad, sinkpad);
if (ret != GST_PAD_LINK_OK) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link rtpsrc to rtpbin");
@ -616,6 +725,19 @@ pipeline_link_elements (CallsSipMediaPipeline *self,
/* can only link to depayloader after RTP payload has been verified */
g_signal_connect (self->rtpbin, "pad-added", G_CALLBACK (on_pad_added), self->depayloader);
/* request-encoder and request-decoder signals have been emitted after linking pads from rtpbin */
if (self->request_rtpbin_rtp_decoder_id)
g_signal_handler_disconnect (self->rtpbin, self->request_rtpbin_rtp_decoder_id);
if (self->request_rtpbin_rtp_encoder_id)
g_signal_handler_disconnect (self->rtpbin, self->request_rtpbin_rtp_encoder_id);
if (self->request_rtpbin_rtcp_decoder_id)
g_signal_handler_disconnect (self->rtpbin, self->request_rtpbin_rtcp_decoder_id);
if (self->request_rtpbin_rtcp_encoder_id)
g_signal_handler_disconnect (self->rtpbin, self->request_rtpbin_rtcp_encoder_id);
return TRUE;
}
@ -657,7 +779,7 @@ pipeline_setup_codecs (CallsSipMediaPipeline *self,
}
/* UDP src capabilities */
caps_string = media_codec_get_gst_capabilities (codec, FALSE);
caps_string = media_codec_get_gst_capabilities (codec, self->use_srtp);
g_debug ("Capabilities:\n%s", caps_string);
caps = gst_caps_from_string (caps_string);
@ -778,6 +900,8 @@ calls_sip_media_pipeline_finalize (GObject *object)
gst_object_unref (self->pipeline);
gst_bus_remove_watch (self->bus);
gst_object_unref (self->bus);
gst_object_unref (self->srtpenc);
gst_object_unref (self->srtpdec);
g_free (self->remote);
@ -854,7 +978,7 @@ usr2_handler (CallsSipMediaPipeline *self)
self->element_map_playing,
self->element_map_paused,
self->element_map_stopped,
EL_ALL_RTP,
self->use_srtp ? EL_ALL_SRTP : EL_ALL_RTP,
self->state);
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (self->pipeline),