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sip: media-pipeline: Fix socket reuse

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We were using two distinct pipelines, one for receiving and one for
sending. The receive pipeline was set to the playing state to allocate
the sockets which we would reuse for the sending direction for our NAT
traversal scheme.

The rework to a single pipeline broke reusing sockets subtly.

This happened because the state of the GstUDPSrc could be reset leading
to newly allocated sockets once the pipeline is set to play.

This is now fixed by locking the state of the GstUDPSrc in the ready
state during socket reuse setup and while the pipeline is paused.

Additionally get rid of the "close-socket" property on the udp sources
because it was never needed.

Fixes aa446f82

sq
This commit is contained in:
Evangelos Ribeiro Tzaras 2022-04-22 17:46:50 +02:00
parent f44b4c7ef8
commit 605776641d
1 changed files with 31 additions and 17 deletions
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48
plugins/sip/calls-sip-media-pipeline.c
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@ -370,8 +370,9 @@ setup_socket_reuse (CallsSipMediaPipeline *self,
g_autoptr (GSocket) rtp_sock = NULL;
g_autoptr (GSocket) rtcp_sock = NULL;
/* Set udp sources to ready to get ports allocated */
gst_element_set_state (self->pipeline, GST_STATE_READY);
/* set rtp element ready and lock it's state so it doesn't get stopped */
gst_element_set_locked_state (self->rtp_src, TRUE);
gst_element_set_state (self->rtp_src, GST_STATE_READY);
g_object_get (self->rtp_src, "used-socket", &rtp_sock, NULL);
if (!rtp_sock) {
@ -381,6 +382,16 @@ setup_socket_reuse (CallsSipMediaPipeline *self,
return FALSE;
}
/* configure socket and don't close it, since it belongs to rtp_src */
g_object_set (self->rtp_sink,
"socket", rtp_sock,
"close-socket", FALSE,
NULL);
/* set rtcp element ready and lock it's state so it doesn't get stopped */
gst_element_set_locked_state (self->rtcp_src, TRUE);
gst_element_set_state (self->rtcp_src, GST_STATE_READY);
g_object_get (self->rtcp_src, "used-socket", &rtcp_sock, NULL);
if (!rtcp_sock) {
if (error)
@ -389,12 +400,8 @@ setup_socket_reuse (CallsSipMediaPipeline *self,
return FALSE;
}
/* Ports are allocated. Let's reuse the socket for rtcp source in the sink for NAT traversal*/
g_object_set (self->rtp_sink,
"socket", rtp_sock,
"close-socket", FALSE,
NULL);
/* configure socket and don't close it, since it belongs to rtcp_src */
g_object_set (self->rtcp_sink,
"socket", rtcp_sock,
"close-socket", FALSE,
@ -478,17 +485,9 @@ pipeline_init (CallsSipMediaPipeline *self,
MAKE_ELEMENT (rtcp_sink, "udpsink", "rtcp-udp-sink");
/* port 0 means letting the OS allocate */
g_object_set (self->rtp_src,
"port", 0,
"close-socket", FALSE,
"reuse", TRUE,
NULL);
g_object_set (self->rtp_src, "port", 0, NULL);
g_object_set (self->rtcp_src,
"port", 0,
"close-socket", FALSE,
"reuse", TRUE,
NULL);
g_object_set (self->rtcp_src, "port", 0, NULL);
g_object_set (self->rtp_sink, "async", FALSE, "sync", FALSE, NULL);
g_object_set (self->rtcp_sink, "async", FALSE, "sync", FALSE, NULL);
@ -993,6 +992,10 @@ calls_sip_media_pipeline_start (CallsSipMediaPipeline *self)
calls_sip_media_pipeline_get_rtp_port (self),
calls_sip_media_pipeline_get_rtcp_port (self));
/* unlock the state of our udp sources, see setup_socket_reuse() */
gst_element_set_locked_state (self->rtp_src, FALSE);
gst_element_set_locked_state (self->rtcp_src, FALSE);
gst_element_set_state (self->pipeline, GST_STATE_PLAYING);
g_debug ("RTP/RTCP port after starting pipeline: %d/%d",
@ -1013,6 +1016,11 @@ calls_sip_media_pipeline_stop (CallsSipMediaPipeline *self)
g_debug ("Stopping media pipeline");
gst_element_set_locked_state (self->rtp_src, FALSE);
gst_element_set_locked_state (self->rtcp_src, FALSE);
gst_element_set_locked_state (self->rtp_sink, FALSE);
gst_element_set_locked_state (self->rtcp_sink, FALSE);
gst_element_set_state (self->pipeline, GST_STATE_NULL);
set_state (self, CALLS_MEDIA_PIPELINE_STATE_STOP_PENDING);
@ -1048,6 +1056,12 @@ calls_sip_media_pipeline_pause (CallsSipMediaPipeline *self,
"Unpausing");
/* leave udpsrc running to prevent timeouts */
gst_element_set_locked_state (self->rtp_src, pause);
gst_element_set_locked_state (self->rtcp_src, pause);
gst_element_set_locked_state (self->rtp_sink, pause);
gst_element_set_locked_state (self->rtcp_sink, pause);
gst_element_set_state (self->pipeline, pause ?
GST_STATE_PAUSED :
GST_STATE_PLAYING);