mirror of
https://github.com/fairwaves/openbts-2.8.git
synced 2025-11-03 21:43:16 +00:00
3d4ec90bd1
control flows. Basically, there was a small bug where I forgot to listen for an OK, which caused transactions to stick around too long and fuck everything up. That was quickly fixed, but my compulsion required me to reorganize the code to avoid such errors in the future. I implemented another new feature as well, we now respond with a 480 Temporarily Unavailable message (rather than cancel) when canceling an INCOMING call. The prior behavior was incorrect. git-svn-id: http://wush.net/svn/range/software/public/openbts/trunk@3264 19bc5d8c-e614-43d4-8b26-e1612bc8e597
1109 lines
32 KiB
C++
1109 lines
32 KiB
C++
/*
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* Copyright 2008 Free Software Foundation, Inc.
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*
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* This software is distributed under the terms of the GNU Affero Public License.
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* See the COPYING file in the main directory for details.
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*
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* This use of this software may be subject to additional restrictions.
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* See the LEGAL file in the main directory for details.
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This program is free software: you can redistribute it and/or modify
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it under the terms of the GNU Affero General Public License as published by
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the Free Software Foundation, either version 3 of the License, or
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(at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU Affero General Public License for more details.
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You should have received a copy of the GNU Affero General Public License
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along with this program. If not, see <http://www.gnu.org/licenses/>.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <signal.h>
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#include <ortp/ortp.h>
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#include <osipparser2/sdp_message.h>
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#include <osipparser2/osip_md5.h>
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#include "SIPInterface.h"
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#include "SIPUtility.h"
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#include "SIPMessage.h"
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using namespace std;
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using namespace SIP;
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#define DEBUG 1
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#define MAX_VIA 10
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osip_message_t * SIP::sip_register( const char * sip_username, short timeout, short wlocal_port, const char * local_ip, const char * proxy_ip, const char * from_tag, const char * via_branch, const char * call_id, int cseq) {
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char local_port[10];
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sprintf(local_port,"%i",wlocal_port);
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// Message URI
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osip_message_t * request;
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osip_message_init(&request);
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// FIXME -- Should use the "force_update" function.
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request->message_property = 2; // buffer is not synchronized with object
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request->sip_method = strdup("REGISTER");
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osip_message_set_version(request, strdup("SIP/2.0"));
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osip_uri_init(&request->req_uri);
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osip_uri_set_host(request->req_uri, strdup(proxy_ip));
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// VIA
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osip_via_t * via;
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osip_via_init(&via);
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via_set_version(via, strdup("2.0"));
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via_set_protocol(via, strdup("UDP"));
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via_set_host(via, strdup(local_ip));
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via_set_port(via, strdup(local_port));
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// VIA BRANCH
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osip_via_set_branch(via, strdup(via_branch));
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// MAX FORWARDS
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osip_message_set_max_forwards(request, strdup(gConfig.getStr("SIP.MaxForwards").c_str()));
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char * via_str;
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osip_via_to_str(via, &via_str);
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osip_message_set_via(request, via_str);
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// FROM
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osip_from_init(&request->from);
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osip_from_set_displayname(request->from, strdup(sip_username));
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// FROM TAG
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osip_from_set_tag(request->from, strdup(from_tag));
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osip_uri_init(&request->from->url);
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osip_uri_set_host(request->from->url, strdup(proxy_ip));
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osip_uri_set_username(request->from->url, strdup(sip_username));
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// TO
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osip_to_init(&request->to);
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osip_to_set_displayname(request->to, strdup(sip_username));
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osip_uri_init(&request->to->url);
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osip_uri_set_host(request->to->url, strdup(proxy_ip));
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osip_uri_set_username(request->to->url, strdup(sip_username));
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// CALL ID
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osip_call_id_init(&request->call_id);
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osip_call_id_set_host(request->call_id, strdup(local_ip));
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osip_call_id_set_number(request->call_id, strdup(call_id));
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// CSEQ
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osip_cseq_init(&request->cseq);
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osip_cseq_set_method(request->cseq, strdup("REGISTER"));
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char temp_buf[14];
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sprintf(temp_buf,"%i",cseq);
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osip_cseq_set_number(request->cseq, strdup(temp_buf));
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// CONTACT
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osip_contact_t * con;
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osip_to_init(&con);
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// CONTACT URI
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osip_uri_init(&con->url);
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osip_uri_set_host(con->url, strdup(local_ip));
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osip_uri_set_port(con->url, strdup(local_port));
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osip_uri_set_username(con->url, strdup(sip_username));
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char numbuf[10];
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sprintf(numbuf,"%d",timeout);
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osip_contact_param_add(con, strdup("expires"), strdup(numbuf) );
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// add contact
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osip_list_add(&request->contacts, con, -1);
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return request;
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}
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osip_message_t * SIP::sip_message( const char * dialed_number, const char * sip_username, short wlocal_port, const char * local_ip, const char * proxy_ip, const char * from_tag, const char * via_branch, const char * call_id, int cseq, const char* message, const char* content_type) {
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char local_port[10];
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sprintf(local_port, "%i", wlocal_port);
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osip_message_t * request;
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osip_message_init(&request);
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// FIXME -- Should use the "force_update" function.
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request->message_property = 2;
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// METHOD
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request->sip_method = strdup("MESSAGE");
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osip_message_set_version(request, strdup("SIP/2.0"));
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// REQ.URI
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osip_uri_init(&request->req_uri);
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osip_uri_set_host(request->req_uri, strdup(proxy_ip));
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osip_uri_set_username(request->req_uri, strdup(dialed_number));
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// VIA
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osip_via_t * via;
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osip_via_init(&via);
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via_set_version(via, strdup("2.0"));
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via_set_protocol(via, strdup("UDP"));
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via_set_host(via, strdup(local_ip));
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via_set_port(via, strdup(local_port));
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osip_via_set_branch(via, strdup(via_branch));
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// MAX FORWARDS
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osip_message_set_max_forwards(request, strdup(gConfig.getStr("SIP.MaxForwards").c_str()));
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// add via
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osip_list_add(&request->vias, via, -1);
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// FROM
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osip_from_init(&request->from);
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osip_from_set_displayname(request->from, strdup(sip_username));
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osip_uri_init(&request->from->url);
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osip_uri_set_host(request->from->url, strdup(proxy_ip));
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osip_uri_set_username(request->from->url, strdup(sip_username));
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// FROM TAG
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osip_from_set_tag(request->from, strdup(from_tag));
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// TO
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osip_to_init(&request->to);
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osip_to_set_displayname(request->to, strdup(dialed_number));
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osip_uri_init(&request->to->url);
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osip_uri_set_host(request->to->url, strdup(proxy_ip));
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osip_uri_set_username(request->to->url, strdup(dialed_number));
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// CALL ID
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osip_call_id_init(&request->call_id);
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osip_call_id_set_host(request->call_id, strdup(local_ip));
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osip_call_id_set_number(request->call_id, strdup(call_id));
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// CSEQ
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osip_cseq_init(&request->cseq);
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osip_cseq_set_method(request->cseq, strdup("MESSAGE"));
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char temp_buf[21];
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sprintf(temp_buf,"%i",cseq);
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osip_cseq_set_number(request->cseq, strdup(temp_buf));
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// Content-Type
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if (content_type)
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{
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// Explicit value provided
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osip_message_set_content_type(request, strdup(content_type));
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} else {
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// Default to text/plain
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osip_message_set_content_type(request, strdup("text/plain"));
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}
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// Content-Length
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sprintf(temp_buf,"%u",strlen(message));
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osip_message_set_content_length(request, strdup(temp_buf));
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// Payload.
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osip_message_set_body(request,message,strlen(message));
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return request;
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}
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osip_message_t * SIP::sip_invite5031(short rtp_port, const char * sip_username, short wlocal_port, const char * local_ip, const char* proxy_ip, const char * from_tag, const char * via_branch, const char * call_id, int cseq, unsigned codec)
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{
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char local_port[10];
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sprintf(local_port, "%i", wlocal_port);
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osip_message_t * request;
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osip_message_init(&request);
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// FIXME -- Should use the "force_update" function.
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request->message_property = 2;
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request->sip_method = strdup("INVITE");
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osip_message_set_version(request, strdup("SIP/2.0"));
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osip_uri_init(&request->req_uri);
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osip_uri_set_scheme(request->req_uri, strdup("sip"));
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osip_uri_set_username(request->req_uri, strdup("sos"));
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osip_uri_set_host(request->req_uri, strdup(proxy_ip));
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// VIA
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osip_via_t * via;
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osip_via_init(&via);
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via_set_version(via, strdup("2.0"));
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via_set_protocol(via, strdup("UDP"));
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via_set_host(via, strdup(local_ip));
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via_set_port(via, strdup(local_port));
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// VIA BRANCH
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osip_via_set_branch(via, strdup(via_branch));
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// MAX FORWARDS
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osip_message_set_max_forwards(request, strdup(gConfig.getStr("SIP.MaxForwards").c_str()));
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// add via
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osip_list_add(&request->vias, via, -1);
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// FROM
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osip_from_init(&request->from);
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osip_from_set_displayname(request->from, strdup(sip_username));
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// FROM TAG
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osip_from_set_tag(request->from, strdup(from_tag));
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osip_uri_init(&request->from->url);
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osip_uri_set_host(request->from->url, strdup(local_ip));
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osip_uri_set_username(request->from->url, strdup(sip_username));
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// TO
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osip_to_init(&request->to);
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osip_to_set_displayname(request->to, strdup(""));
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osip_uri_init(&request->to->url);
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osip_uri_set_host(request->to->url, strdup(gConfig.getStr("Emergency.Destination.Host").c_str()));
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osip_uri_set_username(request->to->url, strdup(gConfig.getStr("Emergency.Destination.User").c_str()));
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// If response, we need a to tag.
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//osip_uri_param_t * to_tag_param;
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//osip_from_get_tag(rsp->to, &to_tag_param);
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// CALL ID
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osip_call_id_init(&request->call_id);
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osip_call_id_set_host(request->call_id, strdup(local_ip));
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osip_call_id_set_number(request->call_id, strdup(call_id));
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// CSEQ
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osip_cseq_init(&request->cseq);
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osip_cseq_set_method(request->cseq, strdup("INVITE"));
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char temp_buf[14];
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sprintf(temp_buf,"%i",cseq);
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osip_cseq_set_number(request->cseq, strdup(temp_buf));
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// CONTACT
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osip_contact_t * con;
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osip_to_init(&con);
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// CONTACT URI
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osip_uri_init(&con->url);
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osip_uri_set_host(con->url, strdup(local_ip));
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osip_uri_set_port(con->url, strdup(local_port));
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osip_uri_set_username(con->url, strdup(sip_username));
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osip_contact_param_add(con, strdup("expires"), strdup("3600") );
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// add contact
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osip_list_add(&request->contacts, con, -1);
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sdp_message_t * sdp;
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sdp_message_init(&sdp);
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sdp_message_v_version_set(sdp, strdup("0"));
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sdp_message_o_origin_set(sdp, strdup(sip_username), strdup("0"),
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strdup("0"), strdup("IN"), strdup("IP4"), strdup(local_ip));
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sdp_message_s_name_set(sdp, strdup("Talk Time"));
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sdp_message_t_time_descr_add(sdp, strdup("0"), strdup("0") );
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sprintf(temp_buf,"%i",rtp_port);
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sdp_message_m_media_add(sdp, strdup("audio"),
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strdup(temp_buf), NULL, strdup("RTP/AVP"));
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sdp_message_c_connection_add
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(sdp, 0, strdup("IN"), strdup("IP4"), strdup(local_ip),NULL, NULL);
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// FIXME -- This should also be inside the switch?
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sdp_message_m_payload_add(sdp,0,strdup("3"));
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switch (codec) {
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case RTPuLaw:
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sdp_message_a_attribute_add(sdp,0,strdup("rtpmap"),strdup("0 PCMU/8000"));
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break;
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case RTPGSM610:
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sdp_message_a_attribute_add(sdp,0,strdup("rtpmap"),strdup("3 GSM/8000"));
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break;
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default: assert(0);
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};
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/*
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* We construct a sdp_message_t, turn it into a string, and then treat it
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* like an osip_body_t. This works, and perhaps is how it is supposed to
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* be done, but in any case we're going to have to do the extra processing
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* to turn it into a string first.
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*/
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char * sdp_str;
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sdp_message_to_str(sdp, &sdp_str);
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osip_message_set_body(request, sdp_str, strlen(sdp_str));
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osip_free(sdp_str);
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osip_message_set_content_type(request, strdup("application/sdp"));
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return request;
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}
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osip_message_t * SIP::sip_invite( const char * dialed_number, short rtp_port, const char * sip_username, short wlocal_port, const char * local_ip, const char * proxy_ip, const char * from_tag, const char * via_branch, const char * call_id, int cseq, unsigned codec) {
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char local_port[10];
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sprintf(local_port, "%i", wlocal_port);
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osip_message_t * request;
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osip_message_init(&request);
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// FIXME -- Should use the "force_update" function.
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request->message_property = 2;
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request->sip_method = strdup("INVITE");
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osip_message_set_version(request, strdup("SIP/2.0"));
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osip_uri_init(&request->req_uri);
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osip_uri_set_host(request->req_uri, strdup(proxy_ip));
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osip_uri_set_username(request->req_uri, strdup(dialed_number));
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// VIA
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osip_via_t * via;
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osip_via_init(&via);
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via_set_version(via, strdup("2.0"));
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via_set_protocol(via, strdup("UDP"));
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via_set_host(via, strdup(local_ip));
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via_set_port(via, strdup(local_port));
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// VIA BRANCH
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osip_via_set_branch(via, strdup(via_branch));
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// MAX FORWARDS
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osip_message_set_max_forwards(request, strdup(gConfig.getStr("SIP.MaxForwards").c_str()));
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// add via
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osip_list_add(&request->vias, via, -1);
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// FROM
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osip_from_init(&request->from);
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osip_from_set_displayname(request->from, strdup(sip_username));
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// FROM TAG
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osip_from_set_tag(request->from, strdup(from_tag));
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osip_uri_init(&request->from->url);
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osip_uri_set_host(request->from->url, strdup(proxy_ip));
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osip_uri_set_username(request->from->url, strdup(sip_username));
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// TO
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osip_to_init(&request->to);
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osip_to_set_displayname(request->to, strdup(""));
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osip_uri_init(&request->to->url);
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osip_uri_set_host(request->to->url, strdup(proxy_ip));
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osip_uri_set_username(request->to->url, strdup(dialed_number));
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// If response, we need a to tag.
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//osip_uri_param_t * to_tag_param;
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//osip_from_get_tag(rsp->to, &to_tag_param);
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// CALL ID
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osip_call_id_init(&request->call_id);
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osip_call_id_set_host(request->call_id, strdup(local_ip));
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osip_call_id_set_number(request->call_id, strdup(call_id));
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// CSEQ
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osip_cseq_init(&request->cseq);
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osip_cseq_set_method(request->cseq, strdup("INVITE"));
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char temp_buf[14];
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sprintf(temp_buf,"%i",cseq);
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osip_cseq_set_number(request->cseq, strdup(temp_buf));
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// CONTACT
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osip_contact_t * con;
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osip_to_init(&con);
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// CONTACT URI
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osip_uri_init(&con->url);
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osip_uri_set_host(con->url, strdup(local_ip));
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osip_uri_set_port(con->url, strdup(local_port));
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osip_uri_set_username(con->url, strdup(sip_username));
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osip_contact_param_add(con, strdup("expires"), strdup("3600") );
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// add contact
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osip_list_add(&request->contacts, con, -1);
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sdp_message_t * sdp;
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sdp_message_init(&sdp);
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sdp_message_v_version_set(sdp, strdup("0"));
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sdp_message_o_origin_set(sdp, strdup(sip_username), strdup("0"),
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strdup("0"), strdup("IN"), strdup("IP4"), strdup(local_ip));
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sdp_message_s_name_set(sdp, strdup("Talk Time"));
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sdp_message_t_time_descr_add(sdp, strdup("0"), strdup("0") );
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sprintf(temp_buf,"%i",rtp_port);
|
|
sdp_message_m_media_add(sdp, strdup("audio"),
|
|
strdup(temp_buf), NULL, strdup("RTP/AVP"));
|
|
sdp_message_c_connection_add
|
|
(sdp, 0, strdup("IN"), strdup("IP4"), strdup(local_ip),NULL, NULL);
|
|
|
|
// FIXME -- This should also be inside the switch?
|
|
sdp_message_m_payload_add(sdp,0,strdup("3"));
|
|
switch (codec) {
|
|
case RTPuLaw:
|
|
sdp_message_a_attribute_add(sdp,0,strdup("rtpmap"),strdup("0 PCMU/8000"));
|
|
break;
|
|
case RTPGSM610:
|
|
sdp_message_a_attribute_add(sdp,0,strdup("rtpmap"),strdup("3 GSM/8000"));
|
|
break;
|
|
default: assert(0);
|
|
};
|
|
|
|
/*
|
|
* We construct a sdp_message_t, turn it into a string, and then treat it
|
|
* like an osip_body_t. This works, and perhaps is how it is supposed to
|
|
* be done, but in any case we're going to have to do the extra processing
|
|
* to turn it into a string first.
|
|
*/
|
|
char * sdp_str;
|
|
sdp_message_to_str(sdp, &sdp_str);
|
|
osip_message_set_body(request, sdp_str, strlen(sdp_str));
|
|
osip_free(sdp_str);
|
|
osip_message_set_content_type(request, strdup("application/sdp"));
|
|
|
|
return request;
|
|
}
|
|
|
|
|
|
// Take the authorization produced by an earlier invite message.
|
|
|
|
osip_message_t * SIP::sip_ack(const char * req_uri, const char * dialed_number, const char * sip_username, short wlocal_port, const char * local_ip, const char * proxy_ip, const osip_from_t *from_header, const osip_to_t* to_header, const char * via_branch, const osip_call_id_t* call_id_header, int cseq) {
|
|
|
|
char local_port[20];
|
|
sprintf(local_port, "%i", wlocal_port);
|
|
|
|
osip_message_t * ack;
|
|
osip_message_init(&ack);
|
|
// FIXME -- Should use the "force_update" function.
|
|
ack->message_property = 2;
|
|
ack->sip_method = strdup("ACK");
|
|
osip_message_set_version(ack, strdup("SIP/2.0"));
|
|
|
|
osip_uri_init(&ack->req_uri);
|
|
|
|
// If we are Acking a BYE message then need to
|
|
// set the req_uri to the owner address thats taken from the 200 Okay.
|
|
if( req_uri == NULL ) {
|
|
osip_uri_set_host(ack->req_uri, strdup(proxy_ip));
|
|
} else {
|
|
osip_uri_set_host(ack->req_uri, strdup(req_uri));
|
|
}
|
|
|
|
osip_uri_set_username(ack->req_uri, strdup(dialed_number));
|
|
|
|
// Via
|
|
osip_via_t *via;
|
|
osip_via_init(&via);
|
|
via_set_version(via, strdup("2.0"));
|
|
via_set_protocol(via, strdup("UDP"));
|
|
via_set_host(via, strdup(local_ip));
|
|
via_set_port(via, strdup(local_port));
|
|
|
|
// VIA BRANCH
|
|
osip_via_set_branch(via, strdup(via_branch));
|
|
|
|
// MAX FORWARDS
|
|
osip_message_set_max_forwards(ack, strdup(gConfig.getStr("SIP.MaxForwards").c_str()));
|
|
|
|
// add via
|
|
osip_list_add(&ack->vias, via, -1);
|
|
|
|
osip_from_init(&ack->from);
|
|
osip_from_set_displayname(ack->from, strdup(sip_username));
|
|
osip_uri_init(&ack->from->url);
|
|
osip_uri_set_host(ack->from->url, strdup(proxy_ip));
|
|
osip_uri_set_username(ack->from->url, strdup(sip_username));
|
|
|
|
// from/to headers
|
|
osip_from_clone(from_header, &ack->from);
|
|
osip_to_clone(to_header, &ack->to);
|
|
|
|
// call id
|
|
osip_call_id_clone(call_id_header, &ack->call_id);
|
|
|
|
osip_cseq_init(&ack->cseq);
|
|
osip_cseq_set_method(ack->cseq, strdup("ACK"));
|
|
char temp_buf[14];
|
|
sprintf(temp_buf, "%i", cseq);
|
|
osip_cseq_set_number(ack->cseq, strdup(temp_buf));
|
|
|
|
return ack;
|
|
}
|
|
|
|
|
|
osip_message_t * SIP::sip_bye(const char * req_uri, const char * dialed_number, const char * sip_username, short wlocal_port, const char * local_ip, const char * proxy_ip, short wproxy_port, const osip_from_t* from_header, const osip_to_t* to_header, const char * via_branch, const osip_call_id_t* call_id_header, int cseq) {
|
|
|
|
// FIXME -- We really need some NULL-value error checking in here.
|
|
|
|
char local_port[10];
|
|
sprintf(local_port,"%i",wlocal_port);
|
|
|
|
char proxy_port[10];
|
|
sprintf(proxy_port,"%i",wproxy_port);
|
|
|
|
osip_message_t * bye;
|
|
osip_message_init(&bye);
|
|
// FIXME -- Should use the "force_update" function.
|
|
bye->message_property = 2;
|
|
bye->sip_method = strdup("BYE");
|
|
osip_message_set_version(bye, strdup("SIP/2.0"));
|
|
|
|
//char o_addr[30];
|
|
//get_owner_ip(okay, o_addr);
|
|
|
|
osip_uri_init(&bye->req_uri);
|
|
osip_uri_set_host(bye->req_uri, strdup(req_uri));
|
|
osip_uri_set_username(bye->req_uri, strdup(dialed_number));
|
|
|
|
osip_via_t * via;
|
|
osip_via_init(&via);
|
|
via_set_version(via, strdup("2.0"));
|
|
via_set_protocol(via, strdup("UDP"));
|
|
via_set_host(via, strdup(local_ip));
|
|
via_set_port(via, strdup(local_port));
|
|
|
|
// via branch + max forwards
|
|
osip_via_set_branch(via, strdup(via_branch));
|
|
osip_message_set_max_forwards(bye, strdup(gConfig.getStr("SIP.MaxForwards").c_str()));
|
|
|
|
// add via
|
|
osip_list_add(&bye->vias, via, -1);
|
|
|
|
// from/to header
|
|
osip_from_clone(from_header, &bye->from);
|
|
osip_to_clone(to_header, &bye->to);
|
|
|
|
// Call Id Header
|
|
osip_call_id_clone(call_id_header, &bye->call_id);
|
|
|
|
// Cseq Number
|
|
osip_cseq_init(&bye->cseq);
|
|
osip_cseq_set_method(bye->cseq, strdup("BYE"));
|
|
char temp_buf[12];
|
|
sprintf(temp_buf,"%i",cseq);
|
|
osip_cseq_set_number(bye->cseq, strdup(temp_buf));
|
|
|
|
// Contact
|
|
osip_contact_t * contact;
|
|
osip_contact_init(&contact);
|
|
osip_contact_set_displayname(contact, strdup(sip_username) );
|
|
osip_uri_init(&contact->url);
|
|
osip_uri_set_host(contact->url, strdup(local_ip));
|
|
osip_uri_set_username(contact->url, strdup(sip_username));
|
|
osip_uri_set_port(contact->url, strdup(local_port));
|
|
|
|
// add contact
|
|
osip_list_add(&bye->contacts, contact, -1);
|
|
|
|
return bye;
|
|
}
|
|
|
|
osip_message_t * SIP::sip_temporarily_unavailable( osip_message_t * invite, const char * host, const char * username, short port)
|
|
{
|
|
|
|
if(invite==NULL){ return NULL;}
|
|
|
|
osip_message_t * unavail;
|
|
osip_message_init(&unavail);
|
|
//clone doesn't work -kurtis
|
|
// FIXME -- Should use the "force_update" function.
|
|
unavail->message_property = 2;
|
|
//header stuff first
|
|
unavail->status_code = 480;
|
|
unavail->reason_phrase = strdup("Temporarily Unavailable");
|
|
osip_message_set_version(unavail, strdup("SIP/2.0"));
|
|
|
|
char local_port[10];
|
|
sprintf(local_port, "%i", port);
|
|
|
|
//uri
|
|
osip_uri_init(&unavail->req_uri);
|
|
osip_uri_set_host(unavail->req_uri, strdup(host));
|
|
osip_uri_set_username(unavail->req_uri, strdup(username));
|
|
osip_uri_set_port(unavail->req_uri, strdup(local_port));
|
|
|
|
//via
|
|
osip_via_t * via;
|
|
char * via_str;
|
|
osip_message_get_via(invite, 0, &via);
|
|
osip_via_to_str(via, &via_str);
|
|
osip_message_set_via(unavail, via_str);
|
|
osip_free(via_str);
|
|
|
|
// MAX FORWARDS
|
|
osip_message_set_max_forwards(unavail, strdup(gConfig.getStr("SIP.MaxForwards").c_str()));
|
|
|
|
// from/to header
|
|
osip_from_clone(invite->from, &unavail->from);
|
|
osip_to_clone(invite->to, &unavail->to);
|
|
|
|
//contact
|
|
osip_contact_t * cont;
|
|
char * cont_str;
|
|
osip_message_get_contact(invite, 0, &cont);
|
|
osip_contact_to_str(cont, &cont_str);
|
|
osip_message_set_contact(unavail, cont_str);
|
|
osip_free(cont_str);
|
|
|
|
// Get Call-ID.
|
|
osip_call_id_clone(invite->call_id, &unavail->call_id);
|
|
|
|
// Get Cseq.
|
|
osip_cseq_t * cseq;
|
|
char * cseq_str;
|
|
cseq = osip_message_get_cseq(invite);
|
|
osip_cseq_to_str(cseq ,&cseq_str);
|
|
osip_message_set_cseq(unavail, cseq_str);
|
|
osip_free(cseq_str);
|
|
|
|
return unavail;
|
|
}
|
|
|
|
/* Cancel a previously sent invite */
|
|
osip_message_t * SIP::sip_cancel( osip_message_t * invite, const char * host, const char * username, short port)
|
|
{
|
|
|
|
if(invite==NULL){ return NULL;}
|
|
|
|
osip_message_t * cancel;
|
|
osip_message_init(&cancel);
|
|
//clone doesn't work -kurtis
|
|
//osip_message_clone(invite, &cancel);
|
|
// FIXME -- Should use the "force_update" function.
|
|
cancel->message_property = 2;
|
|
//header stuff first
|
|
cancel->sip_method = strdup("CANCEL");
|
|
osip_message_set_version(cancel, strdup("SIP/2.0"));
|
|
|
|
char local_port[10];
|
|
sprintf(local_port, "%i", port);
|
|
|
|
//uri
|
|
osip_uri_init(&cancel->req_uri);
|
|
osip_uri_set_host(cancel->req_uri, strdup(host));
|
|
osip_uri_set_username(cancel->req_uri, strdup(username));
|
|
osip_uri_set_port(cancel->req_uri, strdup(local_port));
|
|
|
|
//via
|
|
osip_via_t * via;
|
|
char * via_str;
|
|
osip_message_get_via(invite, 0, &via);
|
|
osip_via_to_str(via, &via_str);
|
|
osip_message_set_via(cancel, via_str);
|
|
osip_free(via_str);
|
|
|
|
// from/to header
|
|
osip_from_clone(invite->from, &cancel->from);
|
|
osip_to_clone(invite->to, &cancel->to);
|
|
|
|
//contact
|
|
osip_contact_t * cont;
|
|
char * cont_str;
|
|
osip_message_get_contact(invite, 0, &cont);
|
|
osip_contact_to_str(cont, &cont_str);
|
|
osip_message_set_contact(cancel, cont_str);
|
|
osip_free(cont_str);
|
|
|
|
// Get Call-ID.
|
|
osip_call_id_clone(invite->call_id, &cancel->call_id);
|
|
|
|
// Get Cseq.
|
|
osip_cseq_t * cseq;
|
|
char * cseq_str;
|
|
cseq = osip_message_get_cseq(invite);
|
|
osip_cseq_to_str(cseq ,&cseq_str);
|
|
osip_message_set_cseq(cancel, cseq_str);
|
|
osip_free(cseq_str);
|
|
|
|
//update message type
|
|
osip_cseq_set_method(cancel->cseq, strdup("CANCEL"));
|
|
|
|
return cancel;
|
|
}
|
|
|
|
osip_message_t * SIP::sip_okay_sdp( osip_message_t * inv, const char * sip_username, const char * local_ip, short wlocal_port, short rtp_port, unsigned audio_codec)
|
|
{
|
|
|
|
// Check for consistency.
|
|
if(inv==NULL){ return NULL;}
|
|
|
|
char local_port[10];
|
|
sprintf(local_port, "%i", wlocal_port);
|
|
// k used for error conditions on various osip operations.
|
|
|
|
osip_message_t * okay;
|
|
osip_message_init(&okay);
|
|
// FIXME -- Should use the "force_update" function.
|
|
okay->message_property = 2;
|
|
|
|
// Set Header stuff.
|
|
okay->status_code = 200;
|
|
okay->reason_phrase = strdup("OK");
|
|
osip_message_set_version(okay, strdup("SIP/2.0"));
|
|
osip_uri_init(&okay->req_uri);
|
|
|
|
// Get Record Route.
|
|
// FIXME -- Should use _clone() routines.
|
|
osip_record_route_t * rr;
|
|
char * rr_str;
|
|
osip_message_get_record_route(inv, 0, &rr);
|
|
osip_record_route_to_str(rr, &rr_str);
|
|
osip_message_set_record_route(okay, rr_str);
|
|
osip_free(rr_str);
|
|
|
|
|
|
// SIP Okay needs to repeat the Via tags from the INVITE Message.
|
|
osip_via_t * via;
|
|
char * via_str;
|
|
osip_message_get_via(inv, 0, &via);
|
|
osip_via_to_str(via, &via_str);
|
|
osip_message_set_via(okay, via_str);
|
|
osip_free(via_str);
|
|
|
|
// from/to header
|
|
osip_from_clone(inv->from, &okay->from);
|
|
osip_to_clone(inv->to, &okay->to);
|
|
|
|
// CONTACT URI
|
|
osip_contact_t * con;
|
|
osip_to_init(&con);
|
|
osip_uri_init(&con->url);
|
|
osip_uri_set_host(con->url, strdup(local_ip));
|
|
osip_uri_set_port(con->url, strdup(local_port));
|
|
osip_uri_set_username(con->url, strdup(sip_username));
|
|
osip_contact_param_add(con, strdup("expires"), strdup("3600") );
|
|
|
|
// add contact
|
|
osip_list_add(&okay->contacts, con, -1);
|
|
|
|
// Get Call-ID.
|
|
osip_call_id_clone(inv->call_id, &okay->call_id);
|
|
|
|
// Get Cseq.
|
|
osip_cseq_t * cseq;
|
|
char * cseq_str;
|
|
cseq = osip_message_get_cseq(inv);
|
|
osip_cseq_to_str(cseq ,&cseq_str);
|
|
osip_message_set_cseq(okay, cseq_str);
|
|
osip_free(cseq_str);
|
|
|
|
// Session Description Protocol.
|
|
sdp_message_t * sdp;
|
|
sdp_message_init(&sdp);
|
|
sdp_message_v_version_set(sdp, strdup("0"));
|
|
sdp_message_o_origin_set(sdp, strdup(sip_username), strdup("0"),
|
|
strdup("0"), strdup("IN"), strdup("IP4"), strdup(local_ip));
|
|
|
|
sdp_message_s_name_set(sdp, strdup("Talk Time"));
|
|
sdp_message_t_time_descr_add(sdp, strdup("0"), strdup("0") );
|
|
char temp_buf[10];
|
|
sprintf(temp_buf,"%i", rtp_port);
|
|
sdp_message_m_media_add(sdp, strdup("audio"),
|
|
strdup(temp_buf), NULL, strdup("RTP/AVP"));
|
|
sdp_message_c_connection_add
|
|
(sdp, 0, strdup("IN"), strdup("IP4"), strdup(local_ip),NULL, NULL);
|
|
|
|
// FIXME -- This should also be inside the switch?
|
|
sdp_message_m_payload_add(sdp,0,strdup("3"));
|
|
switch (audio_codec) {
|
|
case RTPuLaw:
|
|
sdp_message_a_attribute_add(sdp,0,strdup("rtpmap"),strdup("0 PCMU/8000"));
|
|
break;
|
|
case RTPGSM610:
|
|
sdp_message_a_attribute_add(sdp,0,strdup("rtpmap"),strdup("3 GSM/8000"));
|
|
break;
|
|
default: assert(0);
|
|
};
|
|
|
|
char * sdp_str;
|
|
sdp_message_to_str(sdp, &sdp_str);
|
|
osip_message_set_body(okay, sdp_str, strlen(sdp_str));
|
|
osip_free(sdp_str);
|
|
|
|
osip_message_set_content_type(okay, strdup("application/sdp"));
|
|
|
|
return okay;
|
|
}
|
|
|
|
|
|
osip_message_t * SIP::sip_b_okay( osip_message_t * bye )
|
|
{
|
|
// Check for consistency.
|
|
if(bye==NULL){ return NULL;}
|
|
|
|
// k used for error conditions on various osip operations.
|
|
|
|
osip_message_t * okay;
|
|
osip_message_init(&okay);
|
|
// FIXME -- Should use the "force_update" function.
|
|
okay->message_property = 2;
|
|
|
|
// Set Header stuff.
|
|
okay->status_code = 200;
|
|
okay->reason_phrase = strdup("OK");
|
|
osip_message_set_version(okay, strdup("SIP/2.0"));
|
|
osip_uri_init(&okay->req_uri);
|
|
|
|
// SIP Okay needs to repeat the Via tags from the BYE Message.
|
|
osip_via_t * via;
|
|
char * via_str;
|
|
osip_message_get_via(bye, 0, &via);
|
|
osip_via_to_str(via, &via_str);
|
|
osip_message_set_via(okay, via_str);
|
|
osip_free(via_str);
|
|
|
|
// from/to header
|
|
osip_from_clone(bye->from, &okay->from);
|
|
osip_to_clone(bye->to, &okay->to);
|
|
|
|
// Get Call-ID.
|
|
osip_call_id_clone(bye->call_id, &okay->call_id);
|
|
|
|
// Get Cseq.
|
|
osip_cseq_t * cseq;
|
|
char * cseq_str;
|
|
cseq = osip_message_get_cseq(bye);
|
|
osip_cseq_to_str(cseq ,&cseq_str);
|
|
osip_message_set_cseq(okay, cseq_str);
|
|
osip_free(cseq_str);
|
|
|
|
return okay;
|
|
}
|
|
|
|
|
|
osip_message_t * SIP::sip_trying( osip_message_t * invite, const char * sip_username, const char * local_ip )
|
|
{
|
|
osip_message_t * trying;
|
|
osip_message_init(&trying);
|
|
// FIXME -- Should use the "force_update" function.
|
|
trying->message_property = 2;
|
|
|
|
// Set Header stuff.
|
|
trying->status_code = 100;
|
|
trying->reason_phrase = strdup("Trying");
|
|
osip_message_set_version(trying, strdup("SIP/2.0"));
|
|
osip_uri_init(&invite->req_uri); // FIXME? -- Invite rather than trying?
|
|
|
|
// Get Record Route.
|
|
osip_via_t * via;
|
|
char * via_str;
|
|
osip_message_get_via(invite, 0, &via);
|
|
osip_via_to_str(via, &via_str);
|
|
osip_message_set_via(trying, via_str);
|
|
osip_free(via_str);
|
|
|
|
// from/to header
|
|
osip_from_clone(invite->from, &trying->from);
|
|
osip_to_clone(invite->to, &trying->to);
|
|
|
|
// Get Call-ID.
|
|
osip_call_id_clone(invite->call_id, &trying->call_id);
|
|
|
|
// Get Cseq.
|
|
osip_cseq_t * cseq;
|
|
char * cseq_str;
|
|
cseq = osip_message_get_cseq(invite);
|
|
osip_cseq_to_str(cseq ,&cseq_str);
|
|
osip_message_set_cseq(trying, cseq_str);
|
|
osip_free(cseq_str);
|
|
|
|
// CONTACT URI
|
|
osip_contact_t * con;
|
|
osip_to_init(&con);
|
|
osip_uri_init(&con->url);
|
|
osip_uri_set_host(con->url, strdup(local_ip));
|
|
//osip_uri_set_port(con->url, strdup(local_port)); // FIXME ??
|
|
osip_uri_set_username(con->url, strdup(sip_username));
|
|
|
|
// add contact
|
|
osip_list_add(&trying->contacts, con, -1);
|
|
|
|
return trying;
|
|
}
|
|
|
|
|
|
osip_message_t * SIP::sip_ringing( osip_message_t * invite, const char * sip_username, const char *local_ip)
|
|
{
|
|
osip_message_t * ringing;
|
|
osip_message_init(&ringing);
|
|
// FIXME -- Should use the "force_update" function.
|
|
ringing->message_property = 2;
|
|
|
|
// Set Header stuff.
|
|
ringing->status_code = 180;
|
|
ringing->reason_phrase = strdup("Ringing");
|
|
osip_message_set_version(ringing, strdup("SIP/2.0"));
|
|
//osip_uri_init(&invite->req_uri);
|
|
|
|
// Get Record Route.
|
|
osip_via_t * via;
|
|
char * via_str;
|
|
osip_message_get_via(invite, 0, &via);
|
|
osip_via_to_str(via, &via_str);
|
|
osip_message_set_via(ringing, via_str);
|
|
osip_free(via_str);
|
|
|
|
// from/to header
|
|
osip_from_clone(invite->from, &ringing->from);
|
|
osip_to_clone(invite->to, &ringing->to);
|
|
|
|
// Get Call-ID.
|
|
osip_call_id_clone(invite->call_id, &ringing->call_id);
|
|
|
|
// Get Cseq.
|
|
osip_cseq_t * cseq;
|
|
char * cseq_str;
|
|
cseq = osip_message_get_cseq(invite);
|
|
osip_cseq_to_str(cseq ,&cseq_str);
|
|
osip_message_set_cseq(ringing, cseq_str);
|
|
osip_free(cseq_str);
|
|
|
|
// CONTACT URI
|
|
osip_contact_t * con;
|
|
osip_to_init(&con);
|
|
osip_uri_init(&con->url);
|
|
osip_uri_set_host(con->url, strdup(local_ip));
|
|
osip_uri_set_username(con->url, strdup(sip_username));
|
|
|
|
// add contact
|
|
osip_list_add(&ringing->contacts, con, -1);
|
|
|
|
return ringing;
|
|
}
|
|
|
|
|
|
osip_message_t * SIP::sip_okay( osip_message_t * inv, const char * sip_username, const char * local_ip, short wlocal_port)
|
|
{
|
|
|
|
// Check for consistency.
|
|
if(inv==NULL){ return NULL;}
|
|
|
|
char local_port[20];
|
|
sprintf(local_port, "%i", wlocal_port);
|
|
|
|
osip_message_t * okay;
|
|
osip_message_init(&okay);
|
|
// FIXME -- Should use the "force_update" function.
|
|
okay->message_property = 2;
|
|
|
|
// FIXME -- Do we really need all of this string conversion?
|
|
|
|
// Set Header stuff.
|
|
okay->status_code = 200;
|
|
okay->reason_phrase = strdup("OK");
|
|
osip_message_set_version(okay, strdup("SIP/2.0"));
|
|
osip_uri_init(&okay->req_uri);
|
|
|
|
// Get Record Route.
|
|
osip_record_route_t * rr;
|
|
char * rr_str;
|
|
osip_message_get_record_route(inv, 0, &rr);
|
|
osip_record_route_to_str(rr, &rr_str);
|
|
osip_message_set_record_route(okay, rr_str);
|
|
osip_free(rr_str);
|
|
|
|
// SIP Okay needs to repeat the Via tags from the INVITE Message.
|
|
// osip_via_clone(inv->via, &okay->via);
|
|
osip_via_t * via;
|
|
char * via_str;
|
|
osip_message_get_via(inv, 0, &via);
|
|
osip_via_to_str(via, &via_str);
|
|
osip_message_set_via(okay, via_str);
|
|
osip_free(via_str);
|
|
|
|
// from/to header
|
|
osip_from_clone(inv->from, &okay->from);
|
|
osip_to_clone(inv->to, &okay->to);
|
|
|
|
// Get Call-ID.
|
|
osip_call_id_clone(inv->call_id, &okay->call_id);
|
|
|
|
// Get Cseq.
|
|
osip_cseq_t * cseq;
|
|
char * cseq_str;
|
|
cseq = osip_message_get_cseq(inv);
|
|
osip_cseq_to_str(cseq ,&cseq_str);
|
|
osip_message_set_cseq(okay, cseq_str);
|
|
osip_free(cseq_str);
|
|
|
|
return okay;
|
|
}
|
|
|
|
|
|
osip_message_t * SIP::sip_info(unsigned info, const char *dialed_number, short rtp_port, const char * sip_username, short wlocal_port, const char * local_ip, const char * proxy_ip, const char * from_tag, const char * via_branch, const osip_call_id_t *call_id_header, int cseq) {
|
|
|
|
char local_port[10];
|
|
sprintf(local_port, "%i", wlocal_port);
|
|
|
|
osip_message_t * request;
|
|
osip_message_init(&request);
|
|
// FIXME -- Should use the "force_update" function.
|
|
request->message_property = 2;
|
|
request->sip_method = strdup("INFO");
|
|
osip_message_set_version(request, strdup("SIP/2.0"));
|
|
osip_uri_init(&request->req_uri);
|
|
osip_uri_set_host(request->req_uri, strdup(proxy_ip));
|
|
osip_uri_set_username(request->req_uri, strdup(dialed_number));
|
|
|
|
// VIA
|
|
osip_via_t * via;
|
|
osip_via_init(&via);
|
|
via_set_version(via, strdup("2.0"));
|
|
via_set_protocol(via, strdup("UDP"));
|
|
via_set_host(via, strdup(local_ip));
|
|
via_set_port(via, strdup(local_port));
|
|
|
|
// VIA BRANCH
|
|
osip_via_set_branch(via, strdup(via_branch));
|
|
|
|
// add via
|
|
osip_list_add(&request->vias, via, -1);
|
|
|
|
// FROM
|
|
osip_from_init(&request->from);
|
|
osip_from_set_displayname(request->from, strdup(sip_username));
|
|
|
|
// FROM TAG
|
|
osip_from_set_tag(request->from, strdup(from_tag));
|
|
|
|
osip_uri_init(&request->from->url);
|
|
osip_uri_set_host(request->from->url, strdup(proxy_ip));
|
|
osip_uri_set_username(request->from->url, strdup(sip_username));
|
|
|
|
// TO
|
|
osip_to_init(&request->to);
|
|
osip_to_set_displayname(request->to, strdup(""));
|
|
osip_uri_init(&request->to->url);
|
|
osip_uri_set_host(request->to->url, strdup(proxy_ip));
|
|
osip_uri_set_username(request->to->url, strdup(dialed_number));
|
|
|
|
// CALL ID
|
|
osip_call_id_clone(call_id_header, &request->call_id);
|
|
|
|
// CSEQ
|
|
osip_cseq_init(&request->cseq);
|
|
osip_cseq_set_method(request->cseq, strdup("INFO"));
|
|
char temp_buf[21];
|
|
sprintf(temp_buf,"%i",cseq);
|
|
osip_cseq_set_number(request->cseq, strdup(temp_buf));
|
|
|
|
osip_message_set_content_type(request, strdup("application/dtmf-relay"));
|
|
char message[31];
|
|
// FIXME -- This duration should probably come from a config file.
|
|
switch (info) {
|
|
case 11:
|
|
snprintf(message,sizeof(message),"Signal=*\nDuration=200");
|
|
break;
|
|
case 12:
|
|
snprintf(message,sizeof(message),"Signal=#\nDuration=200");
|
|
break;
|
|
default:
|
|
snprintf(message,sizeof(message),"Signal=%i\nDuration=200",info);
|
|
}
|
|
sprintf(temp_buf,"%lu",strlen(message));
|
|
osip_message_set_content_length(request, strdup(temp_buf));
|
|
|
|
// Payload.
|
|
osip_message_set_body(request,message,strlen(message));
|
|
|
|
return request;
|
|
}
|
|
|
|
|
|
|
|
// vim: ts=4 sw=4
|