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Copyright (C) 2000 Simon MORLAT (simon.morlat@free.fr)
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This program is free software; you can redistribute it and/or
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modify it under the terms of the GNU General Public License
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as published by the Free Software Foundation; either version 2
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of the License, or (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 General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program; if not, write to the Free Software
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Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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#include "linphonecore.h"
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/* these are useful messages for the status */
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extern gchar *contacting;
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extern gchar *connected;
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extern gchar *contacted;
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int invite_accepted_cb(OsipDialog *dialog, transaction_t *trn, sip_t *sipmsg, void *data)
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OsipUA *ua=osip_dialog_get_ua(dialog);
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LinphoneCore *lc=(LinphoneCore*)ua->data;
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LinphoneDialogParams *diaparams=(LinphoneDialogParams *)lc->call->data;
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StreamParams *audio_params=&diaparams->audio_params;
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linphone_core_thread_enter(lc);
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lc->vtable.display_status(lc,connected);
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if (lc->ringstream!=NULL){
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ring_stop(lc->ringstream);
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}else err=try_open_dsp(lc);
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/* adjust rtp jitter compensation. It must be at least the latency of the sound card */
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jitt_comp=MAX(lc->sound_conf.latency,lc->rtp_conf.audio_jitt_comp);
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lc->audiostream=audio_stream_start_with_sndcard(diaparams->profile,audio_params->localport,
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audio_params->remoteaddr,audio_params->remoteport,
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audio_params->pt,jitt_comp,lc->sound_conf.sndcard);
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linphone_core_thread_leave(lc);
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int bye_cb(OsipDialog *dialog, transaction_t *trn, sip_t *msg, void *p2)
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OsipUA *ua=osip_dialog_get_ua(dialog);
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LinphoneCore *lc=(LinphoneCore*)ua->data;
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LinphoneDialogParams *diaparams;
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linphone_core_thread_enter(lc);
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linphone_core_lock(lc);
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diaparams=(LinphoneDialogParams*)dialog->data;
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/*stop ringing if necessary*/
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if (lc->ringstream!=NULL) {
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ring_stop(lc->ringstream);
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if (lc->audiostream!=NULL) {
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audio_stream_stop(lc->audiostream);
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lc->vtable.display_status(lc,end);
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if (diaparams!=NULL) linphone_dialog_params_destroy(diaparams);
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restore_sound_daemons();
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linphone_core_thread_leave(lc);
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linphone_core_unlock(lc);
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int faillure_cb(OsipDialog *dialog, transaction_t *trn, sip_t *msg, void *p2)
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int err=((int*)p2)[0];
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char *msg486=_("User is busy.");
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char *msg480=_("User is temporarily unavaillable.");
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char *retrymsg=_("%s. Retry after %i minute(s).");
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char *msg600=_("User does not want to be disturbed.");
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char *msg603=_("Call declined.");
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OsipUA *ua=osip_dialog_get_ua(dialog);
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LinphoneCore *lc=(LinphoneCore*)ua->data;
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LinphoneDialogParams *diaparams;
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linphone_core_lock(lc);
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diaparams=(LinphoneDialogParams*)dialog->data;
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linphone_core_thread_enter(lc);
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lc->vtable.display_status(lc,_("User cannot be found at given address."));
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lc->vtable.display_status(lc,_("Remote user cannot support any of proposed codecs."));
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msg_header_getbyname(msg,"retry-after",0,&retry);
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umsg=g_malloc(strlen(tmpmsg)+strlen(retrymsg)+13);
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sprintf(umsg,retrymsg,tmpmsg,atoi(retry->hvalue)/60);
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lc->vtable.display_message(lc,umsg);
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lc->vtable.display_message(lc,tmpmsg);
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lc->vtable.display_message(lc,msg600);
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lc->vtable.display_status(lc,msg603);
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case -110: /* time out, call leg is lost */
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lc->vtable.display_status(lc,_("Timeout."));
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lc->vtable.display_status(lc,_("Remote host was found but refused connection."));
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reason=msg_getreason(err);
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lc->vtable.display_status(lc,reason);
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else g_message("faillure_cb: %s\n",strerror(-err));
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if (lc->ringstream!=NULL) {
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ring_stop(lc->ringstream);
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restore_sound_daemons();
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if (diaparams!=NULL) linphone_dialog_params_destroy(diaparams);
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linphone_core_thread_leave(lc);
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linphone_core_unlock(lc);
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int invite_cb(OsipDialog *dialog, transaction_t *trn, sip_t *sipmsg, void *p)
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from_t *from=sipmsg->from;
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char *barmesg,*rem_url;
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OsipUA *ua=osip_dialog_get_ua(dialog);
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LinphoneCore *lc=(LinphoneCore*)ua->data;
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linphone_core_lock(lc);
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linphone_core_thread_enter(lc);
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/* first check if we can answer successfully to this invite */
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osip_dialog_respond(dialog,trn,486);
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sdpctx=osip_dialog_get_body_context(dialog,"application/sdp",0);
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/* get the result of the negociation */
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switch(SDP_CONTEXT(sdpctx)->negoc_status)
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g_warning("Error during sdp negociation. Cannot accept incoming call.\n");
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osip_dialog_respond(dialog,trn,606);
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from_2char_without_params(from,&rem_url);
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barmesg=g_strdup_printf("%s %s.",rem_url,contacted);
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else barmesg=g_strdup_printf("Somebody %s.",contacted);
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lc->vtable.display_status(lc,barmesg);
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lc->vtable.inv_recv(lc,rem_url);
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if (try_open_dsp(lc)>0){
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g_message("Starting local ring...");
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lc->ringstream=ring_start(lc->sound_conf.local_ring,2,lc->sound_conf.sndcard);
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osip_dialog_respond(dialog,trn,SDP_CONTEXT(sdpctx)->negoc_status);
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linphone_core_unlock(lc);
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linphone_core_thread_leave(lc);
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int informative_cb(OsipDialog *dialog, transaction_t *trn, sip_t *msg, void *p2)
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int err=((int*)p2)[0];
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OsipUA *ua=osip_dialog_get_ua(dialog);
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LinphoneCore *lc=(LinphoneCore*)ua->data;
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linphone_core_thread_enter(lc);
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linphone_core_lock(lc);
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//g_message("We are in 180 switch of informative_cb()");
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if (try_open_dsp(lc)>0)
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lc->ringstream=ring_start(lc->sound_conf.remote_ring,2,lc->sound_conf.sndcard);
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// this is used by cisco gateways
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case 200: /* for the moment only REGISTER request followed by 200 OK are said to the user*/
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if (lc->sip_conf.reg_conf.expires) lc->vtable.display_status(lc,_("Registration successfull."));
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else lc->vtable.display_status(lc,_("Unregistration successfull."));
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msg_getcontact(msg,0,&contact);
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contact_2char(contact,&ctt);
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}else ctt=sgetcopy(_("(No contact information !)"));
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lc->vtable.display_url(lc,_("User is not reachable at the moment but he invites you\nto contact him using the following alternate resource:"),ctt);
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g_warning("unsupported informative message.\n");
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linphone_core_thread_leave(lc);
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linphone_core_unlock(lc);
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/* these are the SdpHandler callbacks: we are called in to be aware of the content
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of the SDP messages exchanged */
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int set_audio_offer(SdpHandler *sdph,SdpContext *ctx)
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ua=body_handler_get_ua(BODY_HANDLER(sdph));
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lc=(LinphoneCore*)ua->data;
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/* set the nat address in the SDP, if given */
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if (lc->net_conf.nat_address!=NULL && lc->net_conf.use_nat){
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sdp_t *sdp=ctx->offer;
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sfree(sdp->c_connection->c_addr);
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sdp->c_connection->c_addr=sgetcopy(lc->net_conf.nat_address);
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elem=lc->codecs_conf.audio_codecs;
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codec=(PayloadType*) elem->data;
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if (payload_type_usable(codec) && payload_type_enabled(codec)){
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sdp_payload_init(&payload);
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payload.a_rtpmap=g_strdup_printf("%s/%i/1",codec->mime_type,codec->clock_rate);
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payload.pt=rtp_profile_get_payload_number_from_rtpmap(lc->local_profile,payload.a_rtpmap);
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payload.localport=lc->rtp_conf.audio_rtp_port;
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/* when setting speex, add a b=<AS>: to choose bitrate */
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if (strcmp(codec->mime_type,"speex")==0){
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payload.b_as_bandwidth=codec->normal_bitrate/1000;
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sdp_handler_add_audio_payload(sdph,ctx,&payload);
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g_free(payload.a_rtpmap);
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elem=g_list_next(elem);
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/* add telephone-event payload*/
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sdp_payload_init(&payload);
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payload.pt=rtp_profile_get_payload_number_from_mime(lc->local_profile,"telephone-event");
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payload.a_rtpmap="telephone-event/8000";
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payload.a_fmtp="0-11";
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sdp_handler_add_audio_payload(sdph,ctx,&payload);
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int set_video_offer(SdpHandler *sdph,SdpContext *ctx)
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ua=body_handler_get_ua(BODY_HANDLER(sdph));
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lc=(LinphoneCore*)ua->data;
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elem=lc->codecs_conf.video_codecs;
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codec=(PayloadType*) elem->data;
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if (payload_type_usable(codec) && payload_type_enabled(codec)){
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sdp_payload_init(&payload);
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payload.a_rtpmap=g_strdup_printf("%s/%i",codec->mime_type,codec->clock_rate);
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payload.localport=lc->rtp_conf.video_rtp_port;
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sdp_handler_add_video_payload(sdph,ctx,&payload);
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g_free(payload.a_rtpmap);
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elem=g_list_next(elem);
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int payload_is_supported(SdpPayload *payload,RtpProfile *local_profile,RtpProfile *dialog_profile)
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if (payload->a_rtpmap!=NULL){
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localpt=rtp_profile_get_payload_number_from_rtpmap(local_profile,payload->a_rtpmap);
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g_warning("payload has no rtpmap.");
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/* this payload is understood, but does the user want to use it ?? */
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PayloadType *rtppayload;
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rtppayload=rtp_profile_get_payload(local_profile,localpt);
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if (rtppayload==NULL) {
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g_warning("strange error !!");
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if (strcmp(rtppayload->mime_type,"telephone-event")!=0){
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if (!(payload_type_usable(rtppayload) && payload_type_enabled(rtppayload))) {
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g_warning("payload %s is not usable or enabled.",rtppayload->mime_type);
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/* this payload is supported in our local rtp profile, so add it to the dialog rtp
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rtppayload=payload_type_clone(rtppayload);
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rtp_profile_set_payload(dialog_profile,payload->pt,rtppayload);
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/* add to the rtp payload type some other parameters (bandwidth) */
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if (payload->b_as_bandwidth!=0) rtppayload->normal_bitrate=payload->b_as_bandwidth*1000;
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int accept_audio_offer(SdpHandler *sdph,SdpContext *ctx,SdpPayload *payload)
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OsipDialog *dialog=BODY_CONTEXT(ctx)->dialog;
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LinphoneDialogParams *diaparams;
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LinphoneCore *lc=(LinphoneCore*) BODY_HANDLER(sdph)->ua->data;
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RtpProfile *remote_profile;
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StreamParams *params;
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if (dialog->data==NULL){
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diaparams=linphone_dialog_params_new();
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dialog->data=(void*) diaparams;
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}else diaparams=(LinphoneDialogParams*)dialog->data;
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if (diaparams->profile==NULL){
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/* create a remote user agent profile */
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diaparams->profile=remote_profile=rtp_profile_new("remote");
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}else remote_profile=diaparams->profile;
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/* see if this codec is supported in our local rtp profile*/
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supported=payload_is_supported(payload,lc->local_profile,remote_profile);
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g_message("Refusing codec %i (%s)",payload->pt,payload->a_rtpmap);
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params=&diaparams->audio_params;
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if (params->initialized==0){
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/* this is the first codec we may accept*/
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params->localport=payload->localport=lc->rtp_conf.audio_rtp_port;
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params->remoteport=payload->remoteport;
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params->line=payload->line;
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params->pt=payload->pt; /* remember the first payload accepted */
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params->remoteaddr=payload->c_addr;
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params->initialized=1;
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/* refuse all other audio lines*/
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if(params->line!=payload->line) return -1;
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int accept_video_offer(SdpHandler *sdph,SdpContext *ctx,SdpPayload *payload)
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OsipDialog *dialog=BODY_CONTEXT(ctx)->dialog;
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LinphoneDialogParams *diaparams;
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LinphoneCore *lc=(LinphoneCore*) BODY_HANDLER(sdph)->ua->data;
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RtpProfile *remote_profile;
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StreamParams *params;
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if (dialog->data==NULL){
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diaparams=linphone_dialog_params_new();
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dialog->data=(void*) diaparams;
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}else diaparams=(LinphoneDialogParams*)dialog->data;
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if (diaparams->profile==NULL){
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/* create a remote user agent profile */
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diaparams->profile=remote_profile=rtp_profile_new("remote");
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}else remote_profile=diaparams->profile;
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/* see if this codec is supported in our local rtp profile*/
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supported=payload_is_supported(payload,lc->local_profile,remote_profile);
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if (!supported) return -1;
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params=&diaparams->video_params;
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if (params->initialized==0){
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/* this is the first codec we may accept*/
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params->localport=payload->localport=lc->rtp_conf.video_rtp_port;
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params->remoteport=payload->remoteport;
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params->line=payload->line;
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params->pt=payload->pt; /* remember the first payload accepted */
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params->remoteaddr=payload->c_addr;
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params->initialized=1;
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/* refuse all other audio lines*/
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if(params->line!=payload->line) return -1;
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int read_audio_answer(SdpHandler *sdph,SdpContext *ctx,SdpPayload *payload)
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OsipDialog *dialog=BODY_CONTEXT(ctx)->dialog;
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LinphoneDialogParams *diaparams;
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LinphoneCore *lc=(LinphoneCore*) BODY_HANDLER(sdph)->ua->data;
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StreamParams *params;
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if (dialog->data==NULL){
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diaparams=linphone_dialog_params_new();
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dialog->data=(void*) diaparams;
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diaparams->profile=rtp_profile_clone_full(lc->local_profile);
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}else diaparams=(LinphoneDialogParams*)dialog->data;
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/* paranoid check: see if this codec is supported in our local rtp profile*/
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supported=payload_is_supported(payload,lc->local_profile,diaparams->profile);
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g_warning("This remote sip phone did not answer properly to my sdp offer!");
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params=&diaparams->audio_params;
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if (params->initialized==0){
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/* this is the first codec we may accept*/
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params->localport=lc->rtp_conf.audio_rtp_port;
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params->remoteport=payload->remoteport;
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params->line=payload->line;
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params->pt=payload->pt; /* remember the first payload accepted */
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params->remoteaddr=payload->c_addr;
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params->initialized=1;
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int read_video_answer(SdpHandler *sdph,SdpContext *ctx,SdpPayload *payload)
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OsipDialog *dialog=BODY_CONTEXT(ctx)->dialog;
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LinphoneDialogParams *diaparams;
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LinphoneCore *lc=(LinphoneCore*) BODY_HANDLER(sdph)->ua->data;
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StreamParams *params;
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if (dialog->data==NULL){
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diaparams=linphone_dialog_params_new();
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dialog->data=(void*) diaparams;
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diaparams->profile=rtp_profile_clone_full(lc->local_profile);
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}else diaparams=(LinphoneDialogParams*)dialog->data;
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/* paranoid check: see if this codec is supported in our local rtp profile*/
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supported=payload_is_supported(payload,lc->local_profile,diaparams->profile);
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g_warning("This remote sip phone did not answer properly to my sdp offer!");
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params=&diaparams->video_params;
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if (params->initialized==0){
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/* this is the first codec we may accept*/
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params->localport=lc->rtp_conf.video_rtp_port;
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params->remoteport=payload->remoteport;
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params->line=payload->line;
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params->pt=payload->pt; /* remember the first payload accepted */
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params->remoteaddr=payload->c_addr;
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params->initialized=1;
added
removed
coreapi/osipuacb.c
