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sip: media-pipeline: Setup socket reuse for RTP and RTCP during initialization

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Now that initialization is split per pipeline and that the OS handles port
allocation we can move setting up socket reuse into the pipeline initialization
step instead of setting it up when starting the media pipelines.

This makes the calls_sip_media_pipeline_start() method a bit simpler.

We're also now reusing sockets for RTCP.

Closes #315
This commit is contained in:
Evangelos Ribeiro Tzaras 2022-02-28 22:04:23 +01:00
parent a7fcb9c0c0
commit 0e3a07aabf
1 changed files with 60 additions and 18 deletions
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78
plugins/sip/calls-sip-media-pipeline.c
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@ -457,6 +457,8 @@ static gboolean
send_pipeline_init (CallsSipMediaPipeline *self,
GError **error)
{
g_autoptr (GSocket) rtp_sock = NULL;
g_autoptr (GSocket) rtcp_sock = NULL;
const char *env_var;
g_assert (CALLS_SIP_MEDIA_PIPELINE (self));
@ -518,7 +520,36 @@ send_pipeline_init (CallsSipMediaPipeline *self,
self->rtcp_send_sink, "host",
G_BINDING_BIDIRECTIONAL);
/* TODO bind sockets */
/* bind sockets for hole punching scheme */
g_object_get (self->rtp_src, "used-socket", &rtp_sock, NULL);
if (!rtp_sock) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not retrieve used socket from RTP udpsrc element");
return FALSE;
}
g_object_set (self->rtp_sink,
"socket", rtp_sock,
"close-socket", FALSE,
NULL);
g_object_get (self->rtcp_recv_src, "used-socket", &rtcp_sock, NULL);
if (!rtcp_sock) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not retrieve used socket from RTCP udpsrc element");
return FALSE;
}
g_object_set (self->rtcp_send_sink,
"socket", rtcp_sock,
"close-socket", FALSE,
NULL);
g_object_set (self->rtcp_send_src,
"socket", rtcp_sock,
"close-socket", FALSE,
NULL);
gst_bin_add (GST_BIN (self->send_pipeline), self->send_rtpbin);
gst_bin_add_many (GST_BIN (self->send_pipeline), self->rtp_sink,
@ -645,6 +676,7 @@ static gboolean
recv_pipeline_init (CallsSipMediaPipeline *self,
GError **error)
{
g_autoptr (GSocket) rtcp_sock = NULL;
const char *env_var;
g_assert (CALLS_SIP_MEDIA_PIPELINE (self));
@ -712,9 +744,32 @@ recv_pipeline_init (CallsSipMediaPipeline *self,
self->bus_recv = gst_pipeline_get_bus (GST_PIPELINE (self->recv_pipeline));
self->bus_watch_recv = gst_bus_add_watch (self->bus_recv, on_bus_message, self);
g_object_set (self->rtp_src,
"close-socket", FALSE,
"reuse", TRUE,
NULL);
g_object_set (self->rtcp_recv_src,
"close-socket", FALSE,
"reuse", TRUE,
NULL);
/* Set pipeline to ready to get ports allocated */
gst_element_set_state (self->recv_pipeline, GST_STATE_READY);
g_object_get (self->rtcp_recv_src, "used-socket", &rtcp_sock, NULL);
if (!rtcp_sock) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not retrieve used socket from RTCP udpsrc element");
return FALSE;
}
/* Ports are allocated. Let's reuse the socket for rtcp source in the sink for NAT traversal*/
g_object_set (self->rtcp_recv_sink,
"socket", rtcp_sock,
"close-socket", FALSE,
NULL);
return TRUE;
}
@ -1036,7 +1091,6 @@ diagnose_ports_in_use (CallsSipMediaPipeline *self)
void
calls_sip_media_pipeline_start (CallsSipMediaPipeline *self)
{
GSocket *socket;
g_return_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self));
if (self->state != CALLS_MEDIA_PIPELINE_STATE_READY) {
@ -1046,25 +1100,13 @@ calls_sip_media_pipeline_start (CallsSipMediaPipeline *self)
g_debug ("Starting media pipeline");
/* First start the receiver pipeline so that
we may reuse the socket in the sender pipeline */
/* TODO can we do something similar for RTCP? */
gst_element_set_state (self->recv_pipeline, GST_STATE_PLAYING);
g_object_get (self->rtp_src, "used-socket", &socket, NULL);
if (socket) {
g_object_set (self->rtp_sink,
"close-socket", FALSE,
"socket", socket,
NULL);
}
else
g_warning ("Could not get used socket of udpsrc element");
/* Now start the sender pipeline */
gst_element_set_state (self->send_pipeline, GST_STATE_PLAYING);
g_debug ("RTP/RTCP port after starting pipeline: %d/%d",
calls_sip_media_pipeline_get_rtp_port (self),
calls_sip_media_pipeline_get_rtcp_port (self));
set_state (self, CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING);
if (self->debug)