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.
This commit is contained in:
@@ -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);
|
||||
|
||||
Reference in New Issue
Block a user