[calls] sip: media-pipeline: Let the OS allocate sockets for udpsrc
- From: Evangelos Ribeiro Tzaras <devrtz src gnome org>
- To: commits-list gnome org
- Cc:
- Subject: [calls] sip: media-pipeline: Let the OS allocate sockets for udpsrc
- Date: Sun, 6 Mar 2022 00:28:32 +0000 (UTC)
commit 53d6082d64637a0455bfe1a28a412d002d0f8aaa
Author: Evangelos Ribeiro Tzaras <devrtz fortysixandtwo eu>
Date: Mon Feb 28 09:49:43 2022 +0100
sip: media-pipeline: Let the OS allocate sockets for udpsrc
First of we get rid of the bindings between from "lport-rtp" and "lport-rtcp" to
the "port" property of the udpsrc elements. The properties themselves will get
removed a little later as the required changes are rather intrusive and we need
some more infrastructure in place before we can do the switch.
plugins/sip/calls-sip-media-pipeline.c | 57 ++++++++++++++++++++++------------
plugins/sip/calls-sip-media-pipeline.h | 2 ++
2 files changed, 40 insertions(+), 19 deletions(-)
---
diff --git a/plugins/sip/calls-sip-media-pipeline.c b/plugins/sip/calls-sip-media-pipeline.c
index 285b4c43..ca9afc61 100644
--- a/plugins/sip/calls-sip-media-pipeline.c
+++ b/plugins/sip/calls-sip-media-pipeline.c
@@ -295,7 +295,7 @@ send_pipeline_setup_codecs (CallsSipMediaPipeline *self,
return send_pipeline_link_elements (self, error);
}
-/** TODO: we're describing the desired state (not the current state)
+/**
* Prepare a skeleton send pipeline where we can later
* plug the codec specific elements into.
*
@@ -363,10 +363,6 @@ send_pipeline_init (CallsSipMediaPipeline *self,
self->rtp_sink, "host",
G_BINDING_BIDIRECTIONAL);
- g_object_bind_property (self, "lport-rtcp",
- self->rtcp_send_src, "port",
- G_BINDING_BIDIRECTIONAL);
-
g_object_bind_property (self, "rport-rtcp",
self->rtcp_send_sink, "port",
G_BINDING_BIDIRECTIONAL);
@@ -375,6 +371,8 @@ send_pipeline_init (CallsSipMediaPipeline *self,
self->rtcp_send_sink, "host",
G_BINDING_BIDIRECTIONAL);
+ /* TODO bind sockets */
+
gst_bin_add (GST_BIN (self->send_pipeline), self->send_rtpbin);
gst_bin_add_many (GST_BIN (self->send_pipeline), self->rtp_sink,
self->rtcp_send_src, self->rtcp_send_sink, NULL);
@@ -485,17 +483,16 @@ recv_pipeline_setup_codecs (CallsSipMediaPipeline *self,
}
-/** TODO: we're describing the desired state (not the current state)
+/**
* Prepares a skeleton receiver pipeline which can later be
* used to plug codec specific element in.
* This pipeline just consists of (minimally linked) rtpbin
* audio sink and two udpsrc elements, one for RTP and one for RTCP.
*
- * The pipeline will be started and stopped to let the OS allocate
- * sockets for us instead of building and providing GSockets ourselves
- * by hand. These GSockets will later be reused for any outgoing
- * traffic for of our hole punching scheme as a simple NAT traversal
- * technique.
+ * The pipeline will be set ready to let the OS allocate sockets
+ * for us instead of building and providing GSockets ourselves
+ * by hand. These GSockets are reused for any outgoing traffic in our
+ * hole punching scheme as a simple NAT traversal technique.
*/
static gboolean
recv_pipeline_init (CallsSipMediaPipeline *self,
@@ -550,13 +547,9 @@ recv_pipeline_init (CallsSipMediaPipeline *self,
NULL);
- g_object_bind_property (self, "lport-rtp",
- self->rtp_src, "port",
- G_BINDING_BIDIRECTIONAL);
-
- g_object_bind_property (self, "lport-rtcp",
- self->rtcp_recv_src, "port",
- G_BINDING_BIDIRECTIONAL);
+ /* port 0 means allocate */
+ g_object_set (self->rtp_src, "port", 0, NULL);
+ g_object_set (self->rtcp_recv_src, "port", 0, NULL);
g_object_bind_property (self, "rport-rtcp",
self->rtcp_recv_sink, "port",
@@ -570,10 +563,11 @@ recv_pipeline_init (CallsSipMediaPipeline *self,
gst_bin_add_many (GST_BIN (self->recv_pipeline), self->rtp_src,
self->rtcp_recv_src, self->rtcp_recv_sink, NULL);
- /* TODO use temporary bus watch for the initial pipeline start/stop */
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);
+ /* Set pipeline to ready to get ports allocated */
+ gst_element_set_state (self->recv_pipeline, GST_STATE_READY);
return TRUE;
}
@@ -973,4 +967,29 @@ calls_sip_media_pipeline_pause (CallsSipMediaPipeline *self,
}
+int
+calls_sip_media_pipeline_get_rtp_port (CallsSipMediaPipeline *self)
+{
+ int port;
+
+ g_return_val_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self), 0);
+
+ g_object_get (self->rtp_src, "port", &port, NULL);
+
+ return port;
+}
+
+
+int
+calls_sip_media_pipeline_get_rtcp_port (CallsSipMediaPipeline *self)
+{
+ int port;
+
+ g_return_val_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self), 0);
+
+ g_object_get (self->rtcp_recv_src, "port", &port, NULL);
+
+ return port;
+}
+
#undef MAKE_ELEMENT
diff --git a/plugins/sip/calls-sip-media-pipeline.h b/plugins/sip/calls-sip-media-pipeline.h
index fdcc9b07..4cc01152 100644
--- a/plugins/sip/calls-sip-media-pipeline.h
+++ b/plugins/sip/calls-sip-media-pipeline.h
@@ -42,5 +42,7 @@ void calls_sip_media_pipeline_start (CallsSip
void calls_sip_media_pipeline_stop (CallsSipMediaPipeline *self);
void calls_sip_media_pipeline_pause (CallsSipMediaPipeline *self,
gboolean pause);
+int calls_sip_media_pipeline_get_rtp_port (CallsSipMediaPipeline *self);
+int calls_sip_media_pipeline_get_rtcp_port (CallsSipMediaPipeline *self);
G_END_DECLS
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