[calls] sip: media-pipeline: Remove lport-rtp and lport-rtcp property
- From: Evangelos Ribeiro Tzaras <devrtz src gnome org>
- To: commits-list gnome org
- Cc:
- Subject: [calls] sip: media-pipeline: Remove lport-rtp and lport-rtcp property
- Date: Sun, 6 Mar 2022 00:28:32 +0000 (UTC)
commit 849f2986096425944ca496b7797f2bb232e5c13f
Author: Evangelos Ribeiro Tzaras <devrtz fortysixandtwo eu>
Date: Mon Feb 28 11:38:06 2022 +0100
sip: media-pipeline: Remove lport-rtp and lport-rtcp property
We're not setting the desired ports from the outside anymore, but rather
querying the ports that have been allocated by the operating system.
Therefore the lport-rtp and lport-rtcp property have become superfluous and are
being removed. We also adapt to changes outside of the pipeline code.
plugins/sip/calls-sip-call.c | 35 +++++++++-------------------------
plugins/sip/calls-sip-call.h | 4 +---
plugins/sip/calls-sip-media-manager.c | 16 ++++++++++------
plugins/sip/calls-sip-media-manager.h | 6 ++++--
plugins/sip/calls-sip-media-pipeline.c | 32 -------------------------------
plugins/sip/calls-sip-origin.c | 12 +++++++-----
plugins/sip/calls-sip-util.c | 16 +---------------
plugins/sip/calls-sip-util.h | 1 -
tests/test-media.c | 14 +++++++-------
9 files changed, 39 insertions(+), 97 deletions(-)
---
diff --git a/plugins/sip/calls-sip-call.c b/plugins/sip/calls-sip-call.c
index 18bb7f43..8270d561 100644
--- a/plugins/sip/calls-sip-call.c
+++ b/plugins/sip/calls-sip-call.c
@@ -65,8 +65,6 @@ struct _CallsSipCall
char *ip;
- guint lport_rtp;
- guint lport_rtcp;
guint rport_rtp;
guint rport_rtcp;
gchar *remote;
@@ -97,18 +95,6 @@ try_setting_up_media_pipeline (CallsSipCall *self)
calls_sip_media_pipeline_set_codec (self->pipeline, codec);
}
- if (!self->lport_rtp || !self->lport_rtcp || !self->remote ||
- !self->rport_rtp || !self->rport_rtcp)
- return FALSE;
-
- g_debug ("Setting local ports: RTP/RTCP %u/%u",
- self->lport_rtp, self->lport_rtcp);
-
- g_object_set (G_OBJECT (self->pipeline),
- "lport-rtp", self->lport_rtp,
- "lport-rtcp", self->lport_rtcp,
- NULL);
-
g_debug ("Setting remote ports: RTP/RTCP %u/%u",
self->rport_rtp, self->rport_rtcp);
@@ -127,7 +113,7 @@ calls_sip_call_answer (CallsCall *call)
{
CallsSipCall *self;
g_autofree gchar *local_sdp = NULL;
- guint local_port = get_port_for_rtp ();
+ guint rtp_port, rtcp_port;
g_assert (CALLS_IS_CALL (call));
g_assert (CALLS_IS_SIP_CALL (call));
@@ -141,12 +127,15 @@ calls_sip_call_answer (CallsCall *call)
return;
}
- /* TODO get free port by creating GSocket and passing that to the pipeline */
- calls_sip_call_setup_local_media_connection (self, local_port, local_port + 1);
+ rtp_port = calls_sip_media_pipeline_get_rtp_port (self->pipeline);
+ rtcp_port = calls_sip_media_pipeline_get_rtcp_port (self->pipeline);
+
+ calls_sip_call_setup_local_media_connection (self);
local_sdp = calls_sip_media_manager_get_capabilities (self->manager,
self->ip,
- local_port,
+ rtp_port,
+ rtcp_port,
FALSE,
self->codecs);
@@ -317,19 +306,13 @@ calls_sip_call_init (CallsSipCall *self)
/**
* calls_sip_call_setup_local_media_connection:
* @self: A #CallsSipCall
- * @port_rtp: The RTP port on the the local host
- * @port_rtcp: The RTCP port on the local host
*/
void
-calls_sip_call_setup_local_media_connection (CallsSipCall *self,
- guint port_rtp,
- guint port_rtcp)
+calls_sip_call_setup_local_media_connection (CallsSipCall *self)
{
g_return_if_fail (CALLS_IS_SIP_CALL (self));
- self->lport_rtp = port_rtp;
- self->lport_rtcp = port_rtcp;
-
+ /* XXX maybe we can get rid of this completely */
try_setting_up_media_pipeline (self);
}
diff --git a/plugins/sip/calls-sip-call.h b/plugins/sip/calls-sip-call.h
index dcb6329f..5a57dc37 100644
--- a/plugins/sip/calls-sip-call.h
+++ b/plugins/sip/calls-sip-call.h
@@ -45,9 +45,7 @@ void calls_sip_call_setup_remote_media_connection (CallsSi
const char *remote,
guint port_rtp,
guint port_rtcp);
-void calls_sip_call_setup_local_media_connection (CallsSipCall *self,
- guint port_rtp,
- guint port_rtcp);
+void calls_sip_call_setup_local_media_connection (CallsSipCall *self);
void calls_sip_call_activate_media (CallsSipCall *self,
gboolean enabled);
void calls_sip_call_set_state (CallsSipCall *self,
diff --git a/plugins/sip/calls-sip-media-manager.c b/plugins/sip/calls-sip-media-manager.c
index 1a00161b..9d3746d6 100644
--- a/plugins/sip/calls-sip-media-manager.c
+++ b/plugins/sip/calls-sip-media-manager.c
@@ -222,7 +222,8 @@ calls_sip_media_manager_default (void)
char *
calls_sip_media_manager_get_capabilities (CallsSipMediaManager *self,
const char *own_ip,
- guint port,
+ gint rtp_port,
+ gint rtcp_port,
gboolean use_srtp,
GList *supported_codecs)
{
@@ -245,7 +246,7 @@ calls_sip_media_manager_get_capabilities (CallsSipMediaManager *self,
/* media lines look f.e like "audio 31337 RTP/AVP 9 8 0" */
g_string_append_printf (media_line,
- "m=audio %d RTP/%s", port, payload_type);
+ "m=audio %d RTP/%s", rtp_port, payload_type);
for (node = supported_codecs; node != NULL; node = node->next) {
MediaCodecInfo *codec = node->data;
@@ -259,7 +260,7 @@ calls_sip_media_manager_get_capabilities (CallsSipMediaManager *self,
"\r\n");
}
- g_string_append_printf (attribute_lines, "a=rtcp:%d\r\n", port + 1);
+ g_string_append_printf (attribute_lines, "a=rtcp:%d\r\n", rtcp_port);
done:
if (own_ip && *own_ip)
@@ -287,7 +288,8 @@ calls_sip_media_manager_get_capabilities (CallsSipMediaManager *self,
/* calls_sip_media_manager_static_capabilities:
*
* @self: A #CallsSipMediaManager
- * @port: Should eventually come from the ICE stack
+ * @rtp_port: Port to use for RTP. Should eventually come from the ICE stack
+ * @rtcp_port: Port to use for RTCP.Should eventually come from the ICE stack
* @use_srtp: Whether to use srtp (not really handled)
*
* Returns: (transfer full): string describing capabilities
@@ -296,14 +298,16 @@ calls_sip_media_manager_get_capabilities (CallsSipMediaManager *self,
char *
calls_sip_media_manager_static_capabilities (CallsSipMediaManager *self,
const char *own_ip,
- guint port,
+ gint rtp_port,
+ gint rtcp_port,
gboolean use_srtp)
{
g_return_val_if_fail (CALLS_IS_SIP_MEDIA_MANAGER (self), NULL);
return calls_sip_media_manager_get_capabilities (self,
own_ip,
- port,
+ rtp_port,
+ rtcp_port,
use_srtp,
self->preferred_codecs);
}
diff --git a/plugins/sip/calls-sip-media-manager.h b/plugins/sip/calls-sip-media-manager.h
index 076235bc..4158d788 100644
--- a/plugins/sip/calls-sip-media-manager.h
+++ b/plugins/sip/calls-sip-media-manager.h
@@ -40,12 +40,14 @@ G_DECLARE_FINAL_TYPE (CallsSipMediaManager, calls_sip_media_manager, CALLS, SIP_
CallsSipMediaManager* calls_sip_media_manager_default (void);
gchar* calls_sip_media_manager_get_capabilities (CallsSipMediaManager *self,
const char *own_ip,
- guint port,
+ gint rtp_port,
+ gint rtcp_port,
gboolean use_srtp,
GList
*supported_codecs);
gchar* calls_sip_media_manager_static_capabilities (CallsSipMediaManager *self,
const char *own_ip,
- guint port,
+ gint rtp_port,
+ gint rtcp_port,
gboolean use_srtp);
gboolean calls_sip_media_manager_supports_media (CallsSipMediaManager *self,
const char *media_type);
diff --git a/plugins/sip/calls-sip-media-pipeline.c b/plugins/sip/calls-sip-media-pipeline.c
index 4d64310a..9c740b84 100644
--- a/plugins/sip/calls-sip-media-pipeline.c
+++ b/plugins/sip/calls-sip-media-pipeline.c
@@ -95,9 +95,7 @@ enum {
PROP_0,
PROP_CODEC,
PROP_REMOTE,
- PROP_LPORT_RTP,
PROP_RPORT_RTP,
- PROP_LPORT_RTCP,
PROP_RPORT_RTCP,
PROP_DEBUG,
PROP_STATE,
@@ -128,10 +126,8 @@ struct _CallsSipMediaPipeline {
char *remote;
gint rport_rtp;
- gint lport_rtp;
gint rport_rtcp;
- gint lport_rtcp;
/* Gstreamer Elements (sending) */
GstElement *send_pipeline;
@@ -740,14 +736,6 @@ calls_sip_media_pipeline_get_property (GObject *object,
g_value_set_string (value, self->remote);
break;
- case PROP_LPORT_RTP:
- g_value_set_uint (value, self->lport_rtp);
- break;
-
- case PROP_LPORT_RTCP:
- g_value_set_uint (value, self->lport_rtcp);
- break;
-
case PROP_RPORT_RTP:
g_value_set_uint (value, self->rport_rtp);
break;
@@ -789,14 +777,6 @@ calls_sip_media_pipeline_set_property (GObject *object,
self->remote = g_value_dup_string (value);
break;
- case PROP_LPORT_RTP:
- self->lport_rtp = g_value_get_uint (value);
- break;
-
- case PROP_LPORT_RTCP:
- self->lport_rtcp = g_value_get_uint (value);
- break;
-
case PROP_RPORT_RTP:
self->rport_rtp = g_value_get_uint (value);
break;
@@ -884,18 +864,6 @@ calls_sip_media_pipeline_class_init (CallsSipMediaPipelineClass *klass)
NULL,
G_PARAM_READWRITE);
- props[PROP_LPORT_RTP] = g_param_spec_uint ("lport-rtp",
- "lport-rtp",
- "local rtp port",
- 1025, 65535, 5002,
- G_PARAM_READWRITE);
-
- props[PROP_LPORT_RTCP] = g_param_spec_uint ("lport-rtcp",
- "lport-rtcp",
- "local rtcp port",
- 1025, 65535, 5003,
- G_PARAM_READWRITE);
-
props[PROP_RPORT_RTP] = g_param_spec_uint ("rport-rtp",
"rport-rtp",
"remote rtp port",
diff --git a/plugins/sip/calls-sip-origin.c b/plugins/sip/calls-sip-origin.c
index c649d57e..81057f57 100644
--- a/plugins/sip/calls-sip-origin.c
+++ b/plugins/sip/calls-sip-origin.c
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2021 Purism SPC
+ * Copyright (C) 2021-2022 Purism SPC
*
* This file is part of Calls.
*
@@ -239,10 +239,11 @@ add_call (CallsSipOrigin *self,
g_autofree gchar *local_sdp = NULL;
g_auto (GStrv) address_split = NULL;
const char *call_address = address;
+ gint rtp_port, rtcp_port;
- /* TODO get free port by creating GSocket and passing that to the pipeline */
- guint local_port = get_port_for_rtp ();
pipeline = calls_sip_media_manager_get_pipeline (self->media_manager);
+ rtp_port = calls_sip_media_pipeline_get_rtp_port (pipeline);
+ rtcp_port = calls_sip_media_pipeline_get_rtcp_port (pipeline);
if (self->can_tel) {
address_split = g_strsplit_set (address, ":@;", -1);
@@ -271,11 +272,12 @@ add_call (CallsSipOrigin *self,
self);
if (!inbound) {
- calls_sip_call_setup_local_media_connection (sip_call, local_port, local_port + 1);
+ calls_sip_call_setup_local_media_connection (sip_call);
local_sdp = calls_sip_media_manager_static_capabilities (self->media_manager,
self->own_ip,
- local_port,
+ rtp_port,
+ rtcp_port,
FALSE);
g_assert (local_sdp);
diff --git a/plugins/sip/calls-sip-util.c b/plugins/sip/calls-sip-util.c
index d5ea2a39..a4349e21 100644
--- a/plugins/sip/calls-sip-util.c
+++ b/plugins/sip/calls-sip-util.c
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2021 Purism SPC
+ * Copyright (C) 2021-2022 Purism SPC
*
* This file is part of Calls.
*
@@ -61,17 +61,3 @@ protocol_is_valid (const char *protocol)
g_strcmp0 (protocol, "TCP") == 0 ||
g_strcmp0 (protocol, "TLS") == 0;
}
-
-#define RTP_PORT_MIN 20000
-#define RTP_PORT_MAX 65534
-guint
-get_port_for_rtp (void)
-{
- const guint rand_range = RTP_PORT_MAX - RTP_PORT_MIN;
- guint rand = (g_random_int () % rand_range) + RTP_PORT_MIN;
-
- /* RTP ports must be even */
- return rand % 2 == 0 ? rand : rand + 1;
-}
-#undef RTP_PORT_MIN
-#undef RTP_PORT_MAX
diff --git a/plugins/sip/calls-sip-util.h b/plugins/sip/calls-sip-util.h
index a0ebb02a..86b47098 100644
--- a/plugins/sip/calls-sip-util.h
+++ b/plugins/sip/calls-sip-util.h
@@ -62,4 +62,3 @@ gboolean check_sips (const char *addr);
gboolean check_ipv6 (const char *host);
const char *get_protocol_prefix (const char *protocol);
gboolean protocol_is_valid (const char *protocol);
-guint get_port_for_rtp (void);
diff --git a/tests/test-media.c b/tests/test-media.c
index 84793ff1..c02f9654 100644
--- a/tests/test-media.c
+++ b/tests/test-media.c
@@ -48,7 +48,7 @@ test_sip_media_manager_caps (void)
/* PCMA RTP */
sdp_message =
- calls_sip_media_manager_get_capabilities (manager, NULL, 40002, FALSE, codecs);
+ calls_sip_media_manager_get_capabilities (manager, NULL, 40002, 40003, FALSE, codecs);
g_assert_true (sdp_message);
g_assert_true (find_string_in_sdp_message (sdp_message,
@@ -64,7 +64,7 @@ test_sip_media_manager_caps (void)
/* PCMA SRTP */
sdp_message =
- calls_sip_media_manager_get_capabilities (manager, NULL, 42002, TRUE, codecs);
+ calls_sip_media_manager_get_capabilities (manager, NULL, 42002, 42003, TRUE, codecs);
g_assert_true (sdp_message);
g_assert_true (find_string_in_sdp_message (sdp_message,
"m=audio 42002 RTP/SAVP 8"));
@@ -78,7 +78,7 @@ test_sip_media_manager_caps (void)
codecs = g_list_append (NULL, media_codec_by_name ("G722"));
sdp_message =
- calls_sip_media_manager_get_capabilities (manager, NULL, 42042, FALSE, codecs);
+ calls_sip_media_manager_get_capabilities (manager, NULL, 42042, 55543, FALSE, codecs);
g_assert_true (sdp_message);
g_assert_true (find_string_in_sdp_message (sdp_message,
@@ -86,7 +86,7 @@ test_sip_media_manager_caps (void)
g_assert_true (find_string_in_sdp_message (sdp_message,
"a=rtpmap:9 G722/8000"));
g_assert_true (find_string_in_sdp_message (sdp_message,
- "a=rtcp:42043"));
+ "a=rtcp:55543"));
g_clear_pointer (&codecs, g_list_free);
g_free (sdp_message);
@@ -99,7 +99,7 @@ test_sip_media_manager_caps (void)
codecs = g_list_append (codecs, media_codec_by_name ("PCMA"));
sdp_message =
- calls_sip_media_manager_get_capabilities (manager, NULL, 33340, FALSE, codecs);
+ calls_sip_media_manager_get_capabilities (manager, NULL, 33340, 33341, FALSE, codecs);
g_assert_true (sdp_message);
g_assert_true (find_string_in_sdp_message (sdp_message,
@@ -123,7 +123,7 @@ test_sip_media_manager_caps (void)
codecs = g_list_append (codecs, media_codec_by_name ("PCMU"));
sdp_message =
- calls_sip_media_manager_get_capabilities (manager, NULL, 18098, TRUE, codecs);
+ calls_sip_media_manager_get_capabilities (manager, NULL, 18098, 18099, TRUE, codecs);
g_assert_true (sdp_message);
g_assert_true (find_string_in_sdp_message (sdp_message,
@@ -138,7 +138,7 @@ test_sip_media_manager_caps (void)
g_test_expect_message ("CallsSipMediaManager", G_LOG_LEVEL_WARNING,
"No supported codecs found. Can't build meaningful SDP message");
sdp_message =
- calls_sip_media_manager_get_capabilities (manager, NULL, 25048, FALSE, NULL);
+ calls_sip_media_manager_get_capabilities (manager, NULL, 25048, 25049, FALSE, NULL);
g_test_assert_expected_messages ();
g_assert_true (sdp_message);
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