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sip: media-pipeline: Remove lport-rtp and lport-rtcp property

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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.
This commit is contained in:
Evangelos Ribeiro Tzaras 2022-02-28 11:38:06 +01:00
parent aeebdfbf53
commit 849f298609
9 changed files with 39 additions and 97 deletions
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35
plugins/sip/calls-sip-call.c
View file

@ -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);
}

4
plugins/sip/calls-sip-call.h
View file

@ -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,

16
plugins/sip/calls-sip-media-manager.c
View file

@ -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);
}

6
plugins/sip/calls-sip-media-manager.h
View file

@ -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);

32
plugins/sip/calls-sip-media-pipeline.c
View file

@ -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",

12
plugins/sip/calls-sip-origin.c
View file

@ -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);

16
plugins/sip/calls-sip-util.c
View file

@ -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

1
plugins/sip/calls-sip-util.h
View file

@ -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);

14
tests/test-media.c
View file

@ -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);