/* * Copyright 2008 Free Software Foundation, Inc. * * This software is distributed under the terms of the GNU Affero Public License. * See the COPYING file in the main directory for details. * * This use of this software may be subject to additional restrictions. * See the LEGAL file in the main directory for details. This program is free software: you can redistribute it and/or modify it under the terms of the GNU Affero General Public License as published by the Free Software Foundation, either version 3 of the License, or (at your option) any later version. This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU Affero General Public License for more details. You should have received a copy of the GNU Affero General Public License along with this program. If not, see . */ #include #include #include #include #include #include #include "SIPInterface.h" #include "SIPUtility.h" #include "SIPMessage.h" using namespace std; using namespace SIP; #define DEBUG 1 #define MAX_VIA 10 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) { char local_port[10]; sprintf(local_port,"%i",wlocal_port); // Message URI osip_message_t * request; osip_message_init(&request); // FIXME -- Should use the "force_update" function. request->message_property = 2; // buffer is not synchronized with object request->sip_method = strdup("REGISTER"); 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)); // 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(request, strdup("70")); char * via_str; osip_via_to_str(via, &via_str); osip_message_set_via(request, via_str); // set 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(sip_username)); osip_uri_init(&request->to->url); osip_uri_set_host(request->to->url, strdup(proxy_ip)); osip_uri_set_username(request->to->url, strdup(sip_username)); // CALL ID osip_call_id_init(&request->call_id); osip_call_id_set_host(request->call_id, strdup(local_ip)); osip_call_id_set_number(request->call_id, strdup(call_id)); // CSEQ osip_cseq_init(&request->cseq); osip_cseq_set_method(request->cseq, strdup("REGISTER")); char temp_buf[14]; sprintf(temp_buf,"%i",cseq); osip_cseq_set_number(request->cseq, strdup(temp_buf)); // CONTACT osip_contact_t * con; osip_to_init(&con); // CONTACT URI 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)); char numbuf[10]; sprintf(numbuf,"%d",timeout); osip_contact_param_add(con, strdup("expires"), strdup(numbuf) ); // add contact osip_list_add(&request->contacts, con, -1); // If dont need authentication, return. return request; } osip_message_t * SIP::sip_unregister( 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){ char local_port[10]; sprintf(local_port,"%i",wlocal_port); // Message URI osip_message_t * request; osip_message_init(&request); // FIXME -- Should use the "force_update" function. request->message_property = 2; request->sip_method = strdup("REGISTER"); 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)); // 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(request, strdup("70")); // 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(sip_username)); osip_uri_init(&request->to->url); osip_uri_set_host(request->to->url, strdup(proxy_ip)); osip_uri_set_username(request->to->url, strdup(sip_username)); // CALL ID osip_call_id_init(&request->call_id); osip_call_id_set_host(request->call_id, strdup(local_ip)); osip_call_id_set_number(request->call_id, strdup(call_id)); // CSEQ osip_cseq_init(&request->cseq); osip_cseq_set_method(request->cseq, strdup("REGISTER")); char temp_buf[14]; sprintf(temp_buf,"%i",cseq); osip_cseq_set_number(request->cseq, strdup(temp_buf)); // CONTACT osip_contact_t * con; osip_to_init(&con); // CONTACT URI 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("0") ); // add contact osip_list_add(&request->contacts, con, -1); //osip_message_set_contact(request, strdup(" * ")); // If don't need authentication, return. return request; } 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) { 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; // METHOD request->sip_method = strdup("MESSAGE"); osip_message_set_version(request, strdup("SIP/2.0")); // REQ.URI 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)); osip_via_set_branch(via, strdup(via_branch)); // MAX FORWARDS osip_message_set_max_forwards(request, strdup("70")); // add via osip_list_add(&request->vias, via, -1); // FROM osip_from_init(&request->from); osip_from_set_displayname(request->from, strdup(sip_username)); 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)); // FROM TAG osip_from_set_tag(request->from, strdup(from_tag)); // 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_init(&request->call_id); osip_call_id_set_host(request->call_id, strdup(local_ip)); osip_call_id_set_number(request->call_id, strdup(call_id)); // CSEQ osip_cseq_init(&request->cseq); osip_cseq_set_method(request->cseq, strdup("MESSAGE")); 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("text/plain")); 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; } 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) { 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("INVITE"); 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)); // MAX FORWARDS osip_message_set_max_forwards(request, strdup("70")); // 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)); // If response, we need a to tag. //osip_uri_param_t * to_tag_param; //osip_from_get_tag(rsp->to, &to_tag_param); // CALL ID osip_call_id_init(&request->call_id); osip_call_id_set_host(request->call_id, strdup(local_ip)); osip_call_id_set_number(request->call_id, strdup(call_id)); // CSEQ osip_cseq_init(&request->cseq); osip_cseq_set_method(request->cseq, strdup("INVITE")); char temp_buf[14]; sprintf(temp_buf,"%i",cseq); osip_cseq_set_number(request->cseq, strdup(temp_buf)); // CONTACT osip_contact_t * con; osip_to_init(&con); // CONTACT URI 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(&request->contacts, con, -1); 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") ); 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 char * from_tag, const char * to_tag, const char * via_branch, const char * call_id, 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("70")); // 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 TAG osip_from_set_tag(ack->from, strdup(from_tag)); osip_to_init(&ack->to); osip_to_set_displayname(ack->to, strdup("")); osip_uri_init(&ack->to->url); osip_uri_set_host(ack->to->url, strdup(proxy_ip)); osip_uri_set_username(ack->to->url, strdup(dialed_number)); // To Tag if(to_tag) osip_from_set_tag(ack->to, strdup(to_tag)); osip_call_id_init(&ack->call_id); osip_call_id_set_host(ack->call_id, strdup(local_ip)); osip_call_id_set_number(ack->call_id, strdup(call_id)); osip_cseq_init(&ack->cseq); osip_cseq_set_method(ack->cseq, strdup("INVITE")); 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, const char * from_tag, const char * to_tag, const char * via_branch, const char * call_id, int cseq) { char local_port[10]; sprintf(local_port,"%i",wlocal_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(proxy_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("70")); // add via osip_list_add(&bye->vias, via, -1); // from header osip_from_init(&bye->from); osip_from_set_displayname(bye->from, strdup(sip_username)); osip_uri_init(&bye->from->url); osip_uri_set_host(bye->from->url, strdup(proxy_ip)); osip_uri_set_username(bye->from->url, strdup(sip_username)); osip_from_set_tag(bye->from, strdup(from_tag)); // to header osip_to_init(&bye->to); osip_to_set_displayname(bye->to, strdup(sip_username)); osip_uri_init(&bye->to->url); osip_uri_set_host(bye->to->url, strdup(proxy_ip)); osip_uri_set_username(bye->to->url, strdup(dialed_number)); osip_from_set_tag(bye->to, strdup(to_tag)); // Call Id Header osip_call_id_init(&bye->call_id); osip_call_id_set_host(bye->call_id, strdup(local_ip)); osip_call_id_set_number(bye->call_id, strdup(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; } // 200 Okay is generated as a response to a INVITE from a remote client. osip_message_t * SIP::sip_okay( osip_message_t * inv, const char * sip_username, const char * local_ip, short wlocal_port, const char * to_tag, 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); // Get From. osip_from_t * from; char * from_str; from = osip_message_get_from(inv); osip_from_to_str(from, &from_str); osip_message_set_from(okay, from_str); osip_free(from_str); // Get To. osip_to_t * to; char * to_str; to = osip_message_get_to(inv); osip_from_to_str(inv->to, &to_str); osip_message_set_to(okay, to_str); osip_free(to_str); // FIXME - use the make_tag if (to_tag) osip_to_set_tag(okay->to, strdup(to_tag)); // 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_t * call_id; char * call_id_str; call_id = osip_message_get_call_id(inv); osip_call_id_to_str(call_id, &call_id_str); osip_message_set_call_id(okay, call_id_str); osip_free(call_id_str); // 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 INVITE 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); // Get From. osip_from_t * from; char * from_str; from = osip_message_get_from(bye); osip_from_to_str(from, &from_str); osip_message_set_from(okay, from_str); osip_free(from_str); // Get To. osip_to_t * to; char * to_str; to = osip_message_get_to(bye); osip_from_to_str(bye->to, &to_str); osip_message_set_to(okay, to_str); osip_free(to_str); // Get Call-ID. osip_call_id_t * call_id; char * call_id_str; call_id = osip_message_get_call_id(bye); osip_call_id_to_str(call_id, &call_id_str); osip_message_set_call_id(okay, call_id_str); osip_free(call_id_str); // 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); // Get From. osip_from_t * from; char * from_str; from = osip_message_get_from(invite); osip_from_to_str(from, &from_str); osip_message_set_from(trying, from_str); osip_free(from_str); // Get To. osip_to_t * to; char * to_str; to = osip_message_get_to(invite); osip_from_to_str(to, &to_str); osip_message_set_to(trying, to_str); osip_free(to_str); // Get Call-ID. osip_call_id_t * call_id; char * call_id_str; call_id = osip_message_get_call_id(invite); osip_call_id_to_str(call_id, &call_id_str); osip_message_set_call_id(trying, call_id_str); osip_free(call_id_str); // 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, const char * to_tag ) { 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); // Get From. osip_from_t * from; char * from_str; from = osip_message_get_from(invite); osip_from_to_str(from, &from_str); osip_message_set_from(ringing, from_str); osip_free(from_str); // Get To. osip_to_t * to; char * to_str; to = osip_message_get_to(invite); osip_from_to_str(to, &to_str); osip_message_set_to(ringing, to_str); osip_free(to_str); osip_to_set_tag(ringing->to, strdup(to_tag)); // Get Call-ID. osip_call_id_t * call_id; char * call_id_str; call_id = osip_message_get_call_id(invite); osip_call_id_to_str(call_id, &call_id_str); osip_message_set_call_id(ringing, call_id_str); osip_free(call_id_str); // 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; } // 200 Okay is generated as a response to a MESSAGE from a remote client. osip_message_t * SIP::sip_okay_SMS( osip_message_t * inv, const char * sip_username, const char * local_ip, short wlocal_port, const char * to_tag) { // 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. // FIXME -- This is not working. osip_via_t * via; char * via_str; osip_message_get_via(inv, 1, &via); osip_via_to_str(via, &via_str); osip_message_set_via(okay, via_str); osip_free(via_str); // Get From. osip_from_t * from; char * from_str; from = osip_message_get_from(inv); osip_from_to_str(from, &from_str); osip_message_set_from(okay, from_str); osip_free(from_str); // Get To. osip_to_t * to; char * to_str; to = osip_message_get_to(inv); osip_from_to_str(inv->to, &to_str); osip_message_set_to(okay, to_str); osip_free(to_str); // To tag - NO. DO NOT USE. //if (to_tag!=NULL) osip_to_set_tag(okay->to, strdup(to_tag)); // Get Call-ID. osip_call_id_t * call_id; char * call_id_str; call_id = osip_message_get_call_id(inv); osip_call_id_to_str(call_id, &call_id_str); osip_message_set_call_id(okay, call_id_str); osip_free(call_id_str); // 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 char * call_id, 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_init(&request->call_id); osip_call_id_set_host(request->call_id, strdup(local_ip)); osip_call_id_set_number(request->call_id, strdup(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