mirror of
https://github.com/RangeNetworks/openbts.git
synced 2025-11-01 12:23:34 +00:00
sync of openbts
git-svn-id: http://wush.net/svn/range/software/public/openbts/trunk@6168 19bc5d8c-e614-43d4-8b26-e1612bc8e597
This commit is contained in:
Kurtis Heimerl
@@ -1,24 +1,14 @@
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/*
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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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* This software is distributed under multiple licenses; see the COPYING file in the main directory for licensing information for this specific distribuion.
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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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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.
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*/
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@@ -46,8 +36,8 @@ using namespace SIP;
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void openbts_message_init(osip_message_t ** msg){
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osip_message_init(msg);
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//I think it's like 40 characters
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char tag[60];
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sprintf(tag, "OpenBTS %s Build Date %s", VERSION, __DATE__);
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static const char* userAgent = "OpenBTS " VERSION " Build Date " __DATE__;
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const char *tag = userAgent;
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osip_message_set_user_agent(*msg, strdup(tag));
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}
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@@ -170,7 +160,7 @@ int openbts_message_set_rr(osip_message_t *response, osip_message_t *orig)
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return MSG_NO_ERROR;
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}
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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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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, string *RAND, const char *IMSI, const char *SRES) {
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char local_port[10];
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sprintf(local_port,"%i",wlocal_port);
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@@ -250,6 +240,17 @@ osip_message_t * SIP::sip_register( const char * sip_username, short timeout, sh
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// add contact
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osip_list_add(&request->contacts, con, -1);
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if (SRES) {
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// add authentication
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osip_authorization_t *auth;
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osip_authorization_init(&auth);
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osip_authorization_set_auth_type(auth, osip_strdup("Digest"));
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osip_authorization_set_nonce(auth, osip_strdup(RAND->c_str()));
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osip_authorization_set_uri(auth, osip_strdup(IMSI));
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osip_authorization_set_response(auth, osip_strdup(SRES));
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osip_list_add(&request->authorizations, auth, -1);
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}
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return request;
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}
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@@ -328,7 +329,7 @@ osip_message_t * SIP::sip_message( const char * dialed_number, const char * sip_
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}
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// Content-Length
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sprintf(temp_buf,"%u", static_cast<unsigned>(strlen(message)));
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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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@@ -337,128 +338,6 @@ osip_message_t * SIP::sip_message( const char * dialed_number, const char * sip_
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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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openbts_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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openbts_message_set_sdp(request, sdp);
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// TODO: In the very unlikely event that sdp_str is null, we should probably do some nice cleanup.
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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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@@ -573,7 +452,147 @@ osip_message_t * SIP::sip_invite( const char * dialed_number, short rtp_port, co
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* to turn it into a string first.
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*/
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openbts_message_set_sdp(request, sdp);
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// TODO: In the very unlikely event that sdp_str is null, we should probably do some nice cleanup.
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// TODO: In the unlikely event that sdp_str is null, we should probably do some nice cleanup.
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return request;
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}
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osip_message_t * SIP::sip_reinvite(
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const char* RemoteUsername,
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const char* RemoteIP,
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const char* SIPDisplayname,
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const char* SIPUsername,
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const char* fromTag,
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const char* fromUsername,
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const char* fromIP,
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const char* toTag,
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const char* toUsername,
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const char* toIP,
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const char* viaBranch,
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const char* callID,
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const char* callIP,
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int cseq,
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unsigned codec,
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short wRTPPort,
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const char* SessionID,
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const char* SessionVersion)
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{
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char RTPPort[10];
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sprintf(RTPPort, "%i", wRTPPort);
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const char * BS2port = gConfig.getStr("SIP.Local.Port").c_str();
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const char * BS2IP = gConfig.getStr("SIP.Local.IP").c_str();
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osip_message_t * request;
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osip_message_init(&request);
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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(RemoteIP));
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osip_uri_set_username(request->req_uri, strdup(RemoteUsername));
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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(BS2IP));
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via_set_port(via, strdup(BS2port));
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// VIA BRANCH
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osip_via_set_branch(via, strdup(viaBranch));
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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(fromDisplayname));
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osip_from_set_tag(request->from, strdup(fromTag));
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osip_uri_init(&request->from->url);
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osip_uri_set_host(request->from->url, strdup(fromIP));
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osip_uri_set_username(request->from->url, strdup(fromUsername));
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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(toDisplayname));
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osip_to_set_tag(request->to, strdup(toTag));
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osip_uri_init(&request->to->url);
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osip_uri_set_host(request->to->url, strdup(toIP));
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osip_uri_set_username(request->to->url, strdup(toUsername));
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// CALL ID
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osip_call_id_init(&request->call_id);
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if (*callIP) osip_call_id_set_host(request->call_id, strdup(callIP));
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osip_call_id_set_number(request->call_id, strdup(callID));
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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(BS2IP));
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osip_uri_set_port(con->url, strdup(BS2port));
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osip_uri_set_username(con->url, strdup(SIPUsername));
|
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osip_contact_param_add(con, strdup("expires"), strdup("3600") );
|
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|
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// add contact
|
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osip_list_add(&request->contacts, con, -1);
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|
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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(SIPDisplayname), strdup(SessionID),
|
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strdup(SessionVersion), strdup("IN"), strdup("IP4"), strdup(BS2IP));
|
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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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|
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sdp_message_m_media_add(sdp, strdup("audio"),
|
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strdup(RTPPort), NULL, strdup("RTP/AVP"));
|
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sdp_message_c_connection_add
|
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(sdp, 0, strdup("IN"), strdup("IP4"), strdup(BS2IP),NULL, NULL);
|
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|
||||
// FIXME -- This should also be inside the switch?
|
||||
sdp_message_m_payload_add(sdp,0,strdup("3"));
|
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switch (codec) {
|
||||
case RTPuLaw:
|
||||
sdp_message_a_attribute_add(sdp,0,strdup("rtpmap"),strdup("0 PCMU/8000"));
|
||||
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;
|
||||
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.
|
||||
*/
|
||||
openbts_message_set_sdp(request, sdp);
|
||||
// TODO: In the unlikely event that sdp_str is null, we should probably do some nice cleanup.
|
||||
|
||||
return request;
|
||||
}
|
||||
@@ -645,7 +664,7 @@ osip_message_t * SIP::sip_ack(const char * req_uri, const char * dialed_number,
|
||||
}
|
||||
|
||||
|
||||
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) {
|
||||
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.
|
||||
|
||||
@@ -758,7 +777,7 @@ osip_message_t * SIP::sip_error(osip_message_t * invite, const char * host, con
|
||||
return unavail;
|
||||
}
|
||||
|
||||
/* Cancel a previously sent invite */
|
||||
/* Cancel a previous invite */
|
||||
osip_message_t * SIP::sip_cancel( osip_message_t * invite, const char * host, const char * username, short port)
|
||||
{
|
||||
|
||||
@@ -767,7 +786,7 @@ osip_message_t * SIP::sip_cancel( osip_message_t * invite, const char * host, c
|
||||
osip_message_t * cancel;
|
||||
openbts_message_init(&cancel);
|
||||
//clone doesn't work -kurtis
|
||||
//osip_message_clone(invite, &cancel);
|
||||
//osip_message_clone(invite, &cancel)
|
||||
// FIXME -- Should use the "force_update" function.
|
||||
cancel->message_property = 2;
|
||||
//header stuff first
|
||||
@@ -939,7 +958,7 @@ osip_message_t * SIP::sip_trying( osip_message_t * invite, const char * sip_user
|
||||
|
||||
// Get Via
|
||||
openbts_message_set_via(trying, invite);
|
||||
|
||||
|
||||
// from/to header
|
||||
osip_from_clone(invite->from, &trying->from);
|
||||
osip_to_clone(invite->to, &trying->to);
|
||||
@@ -1005,7 +1024,7 @@ osip_message_t * SIP::sip_ringing( osip_message_t * invite, const char * sip_use
|
||||
}
|
||||
|
||||
|
||||
osip_message_t * SIP::sip_okay( osip_message_t * inv, const char* /*sip_username*/, const char* /*local_ip*/, short wlocal_port)
|
||||
osip_message_t * SIP::sip_okay( osip_message_t * inv, const char * sip_username, const char * local_ip, short wlocal_port)
|
||||
{
|
||||
|
||||
// Check for consistency.
|
||||
@@ -1048,7 +1067,7 @@ osip_message_t * SIP::sip_okay( osip_message_t * inv, const char* /*sip_username
|
||||
}
|
||||
|
||||
|
||||
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) {
|
||||
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);
|
||||
|
||||
Reference in New Issue
Block a user