mirror of
https://github.com/servalproject/serval-dna.git
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1452 lines
47 KiB
C
1452 lines
47 KiB
C
/*
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Serval Voice Over Mesh Protocol (VoMP)
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Copyright (C) 2012 Paul Gardner-Stephen
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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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
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*/
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/*
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VoMP works using a 6-state model of a phone call, and relies on MDP for
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auth-cryption of frames. VoMP provides it's own replay protection.
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*/
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#include "serval.h"
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/* Although we only support one call at a time, we allow for multiple call states.
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This is partly to deal with denial of service attacks that might occur by causing
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the ejection of newly allocated session numbers before the caller has had a chance
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to progress the call to a further state. */
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int vomp_call_count=0;
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int vomp_active_call=-1;
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vomp_call_state vomp_call_states[VOMP_MAX_CALLS];
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struct profile_total vomp_stats;
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int dump_vomp_status();
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void vomp_process_tick(struct sched_ent *alarm);
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/* which codecs we support (set by registered listener) */
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unsigned char vomp_local_codec_list[256];
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/* Now keep track of who wants to know what we are up to */
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int vomp_interested_usock_count=0;
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#define VOMP_MAX_INTERESTED 128
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struct sockaddr_un *vomp_interested_usocks[VOMP_MAX_INTERESTED];
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int vomp_interested_usock_lengths[VOMP_MAX_INTERESTED];
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unsigned long long vomp_interested_expiries[VOMP_MAX_INTERESTED];
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vomp_call_state *vomp_find_call_by_session(int session_token)
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{
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int i;
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for(i=0;i<vomp_call_count;i++)
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if (session_token==vomp_call_states[i].local.session)
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return &vomp_call_states[i];
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return NULL;
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}
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int vomp_generate_session_id(){
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int session_id=0;
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while (!session_id)
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{
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if (urandombytes((unsigned char *)&session_id,sizeof(int)))
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return WHY("Insufficient entropy");
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session_id&=VOMP_SESSION_MASK;
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DEBUGF("session=0x%08x\n",session_id);
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int i;
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/* reject duplicate call session numbers */
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for(i=0;i<vomp_call_count;i++)
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if (session_id==vomp_call_states[i].local.session
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||session_id==vomp_call_states[i].local.session){
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session_id=0;
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break;
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}
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}
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return session_id;
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}
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vomp_call_state *vomp_create_call(unsigned char *remote_sid,
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unsigned char *local_sid,
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unsigned int remote_session,
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unsigned int local_session,
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int remote_state,
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int local_state){
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int i;
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if (!local_session)
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local_session=vomp_generate_session_id();
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vomp_call_state *call = &vomp_call_states[vomp_call_count];
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vomp_call_count++;
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/* prepare slot */
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bzero(call,sizeof(vomp_call_state));
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bcopy(local_sid,call->local.sid,SID_SIZE);
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bcopy(remote_sid,call->remote.sid,SID_SIZE);
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call->local.session=local_session;
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call->remote.session=remote_session;
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call->local.state=local_state;
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call->remote.state=remote_state;
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call->last_sent_status=-1;
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call->create_time=overlay_gettime_ms();
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call->last_activity=call->create_time;
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// fill sample cache with invalid times
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for (i=0;i<VOMP_MAX_RECENT_SAMPLES *4;i++)
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call->seen_samples[i]=0xFFFFFFFF;
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call->alarm.alarm = call->create_time+VOMP_CALL_STATUS_INTERVAL;
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call->alarm.function = vomp_process_tick;
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vomp_stats.name="vomp_process_tick";
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call->alarm.stats=&vomp_stats;
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schedule(&call->alarm);
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WHYF("Returning new call #%d",local_session);
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return call;
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}
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vomp_call_state *vomp_find_or_create_call(unsigned char *remote_sid,
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unsigned char *local_sid,
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unsigned int sender_session,
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unsigned int recvr_session,
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int sender_state,
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int recvr_state)
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{
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int i;
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vomp_call_state *call;
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if (0) printf("%d calls already in progress.\n",vomp_call_count);
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for(i=0;i<vomp_call_count;i++)
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{
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call = &vomp_call_states[i];
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/* do the fast comparison first, and only if that matches proceed to
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the slower SID comparisons */
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if (0)
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fprintf(stderr,"asking for %06x:%06x, this call %06x:%06x\n",
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sender_session,recvr_session,
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call->remote.session,
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call->local.session);
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int checked=0;
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if (call->remote.session&&sender_session) {
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checked++;
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if(sender_session!=call->remote.session)
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continue;
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}
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if (call->local.session&&recvr_session) {
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checked++;
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if(recvr_session!=call->local.session)
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continue;
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}
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if (!checked) continue;
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if (memcmp(remote_sid,call->remote.sid,SID_SIZE)) continue;
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if (memcmp(local_sid,call->local.sid,SID_SIZE)) continue;
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/* it matches. */
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/* Record session number if required */
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if (!call->remote.session)
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call->remote.session=sender_session;
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if (0) {
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WHYF("Returning existing call #%d",i);
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fprintf(stderr,"%06x:%06x matches call #%d %06x:%06x\n",
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sender_session,recvr_session,i,
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call->remote.session,
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call->local.session);
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}
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return call;
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}
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/* Don't create a call record if either party has ended it */
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if (sender_state==VOMP_STATE_CALLENDED || recvr_state==VOMP_STATE_CALLENDED)
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return NULL;
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/* Only create a call record if either party is in CALLPREP state */
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if (sender_state==VOMP_STATE_CALLPREP || recvr_state==VOMP_STATE_CALLPREP)
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return vomp_create_call(remote_sid, local_sid, sender_session, recvr_session, VOMP_STATE_NOCALL, VOMP_STATE_NOCALL);
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return NULL;
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}
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/* send updated call status to end-point and to any interested listeners as
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appropriate */
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int vomp_send_status_remote_audio(vomp_call_state *call, int audio_codec, char *audio, int audio_length)
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{
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overlay_mdp_frame mdp;
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bzero(&mdp,sizeof(mdp));
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mdp.packetTypeAndFlags=MDP_TX;
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bcopy(call->local.sid,mdp.out.src.sid,SID_SIZE);
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mdp.out.src.port=MDP_PORT_VOMP;
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bcopy(call->remote.sid,mdp.out.dst.sid,SID_SIZE);
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mdp.out.dst.port=MDP_PORT_VOMP;
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mdp.out.payload[0]=0x01; /* Normal VoMP frame */
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mdp.out.payload[1]=(call->remote.state<<4)|call->local.state;
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mdp.out.payload[2]=(call->remote.sequence>>8)&0xff;
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mdp.out.payload[3]=(call->remote.sequence>>0)&0xff;
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mdp.out.payload[4]=(call->local.sequence>>8)&0xff;
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mdp.out.payload[5]=(call->local.sequence>>0)&0xff;
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unsigned long long call_millis=overlay_gettime_ms()-call->create_time;
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mdp.out.payload[6]=(call_millis>>8)&0xff;
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mdp.out.payload[7]=(call_millis>>0)&0xff;
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mdp.out.payload[8]=(call->remote.session>>16)&0xff;
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mdp.out.payload[9]=(call->remote.session>>8)&0xff;
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mdp.out.payload[10]=(call->remote.session>>0)&0xff;
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mdp.out.payload[11]=(call->local.session>>16)&0xff;
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mdp.out.payload[12]=(call->local.session>>8)&0xff;
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mdp.out.payload[13]=(call->local.session>>0)&0xff;
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mdp.out.payload_length=14;
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if (call->local.state<VOMP_STATE_RINGINGOUT
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&&call->remote.state<VOMP_STATE_RINGINGOUT){
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/* Also include list of supported codecs */
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if (0) WHY("Including codec list");
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int i;
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for(i=0;i<256;i++)
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if (vomp_local_codec_list[i]) {
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mdp.out.payload[mdp.out.payload_length++]=i;
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if (0) WHYF(" I support the %s codec",vomp_describe_codec(i));
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}
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mdp.out.payload[mdp.out.payload_length++]=0;
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if (0) WHYF("mdp frame with codec list is %d bytes",mdp.out.payload_length);
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}
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if (call->local.state==VOMP_STATE_INCALL && audio && audio_length && vomp_sample_size(audio_codec)==audio_length) {
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unsigned short *len=&mdp.out.payload_length;
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unsigned char *p=&mdp.out.payload[0];
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// WHY("Including audio sample block");
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/* record sample in recent list.
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XXX - What timestamp to attach to the sample?
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Two obvious choices:
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1. The sample is for the most recent n milliseconds; or
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2. The sample is for the next n milliseconds following the
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last sample.
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Option 1 introduces all sorts of problems with sample production
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jitter, where as option 2 has no such problems, but simply requires the
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producer of audio to ensure that they provide exactly the right amount
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of audio, or risk the call getting out of sync. This is a fairly
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reasonable expectation, or else things go to pot.
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Note that in-call slew is the responsibility of the player, not the
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recorder of audio. Basically if the audio queue starts to bank up,
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then the player needs to drop samples.
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*/
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vomp_sample_block *sb=call->recent_samples;
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int rotor=call->recent_sample_rotor%VOMP_MAX_RECENT_SAMPLES;
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sb[rotor].codec=audio_codec;
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sb[rotor].endtime=call->audio_clock+vomp_codec_timespan(sb[rotor].codec)-1;
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sb[rotor].starttime=call->audio_clock;
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call->audio_clock=sb[rotor].endtime+1;
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bcopy(audio,&sb[rotor].bytes[0],audio_length);
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/* write the sample end-time in milliseconds since call establishment */
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p[(*len)++]=(call->audio_clock>>24)&0xff;
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p[(*len)++]=(call->audio_clock>>16)&0xff;
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p[(*len)++]=(call->audio_clock>>8)&0xff;
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p[(*len)++]=(call->audio_clock>>0)&0xff;
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/* stuff frame with most recent sample blocks as a form of preemptive
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retransmission. But don't make the packets too large. */
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while (((*len)+1+audio_length)
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<VOMP_STUFF_BYTES) {
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p[(*len)++]=sb[rotor].codec;
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bcopy(&sb[rotor].bytes[0],&p[*len],vomp_sample_size(sb[rotor].codec));
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(*len)+=vomp_sample_size(sb[rotor].codec);
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rotor--; if (rotor<0) rotor+=VOMP_MAX_RECENT_SAMPLES;
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rotor%=VOMP_MAX_RECENT_SAMPLES;
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// stop if we've run out of samples before we ran out of bytes
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if ((!sb[rotor].endtime)||(sb[rotor].endtime+1==call->audio_clock)) break;
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}
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call->recent_sample_rotor++;
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call->recent_sample_rotor%=VOMP_MAX_RECENT_SAMPLES;
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}
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/* XXX Here we act as our own client. This used to be able to block.
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We should really refactor overlay_mdp_poll() so that we can deliver
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the frame directly.
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Make sure that we don't want (just drop the message if there is
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congestion) */
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overlay_mdp_dispatch(&mdp,1,NULL,0);
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call->local.sequence++;
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return 0;
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}
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int vomp_send_status_remote(vomp_call_state *call){
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return vomp_send_status_remote_audio(call, 0, NULL, 0);
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}
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int vomp_send_mdp_status_audio(vomp_call_state *call, int audio_codec, unsigned int start_time, unsigned int end_time, char *audio, int audio_length){
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if (audio && audio_length && vomp_sample_size(audio_codec)!=audio_length)
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return WHY("Audio frame is the wrong length");
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overlay_mdp_frame mdp;
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DEBUG("Sending mdp client packet");
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bzero(&mdp,sizeof(mdp));
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mdp.packetTypeAndFlags=MDP_VOMPEVENT;
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mdp.vompevent.call_session_token=call->local.session;
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mdp.vompevent.last_activity=call->last_activity;
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if (call->ringing) mdp.vompevent.flags|=VOMPEVENT_RINGING;
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if (call->local.state==VOMP_STATE_CALLENDED)
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mdp.vompevent.flags|=VOMPEVENT_CALLENDED;
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if (call->remote.state==VOMPEVENT_CALLENDED)
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mdp.vompevent.flags|=VOMPEVENT_CALLREJECT;
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if (call->audio_started)
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mdp.vompevent.flags|=VOMPEVENT_AUDIOSTREAMING;
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// TODO ???
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//mdp.vompevent.flags|=VOMPEVENT_CALLCREATED;
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mdp.vompevent.local_state=call->local.state;
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mdp.vompevent.remote_state=call->remote.state;
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bcopy(&call->remote_codec_list[0],&mdp.vompevent.supported_codecs[0],256);
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if (audio && audio_length) {
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if (0) WHYF("Frame contains audio (codec=%s)",
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vomp_describe_codec(audio_codec));
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bcopy(audio,
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&mdp.vompevent.audio_bytes[0],
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audio_length);
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mdp.vompevent.audio_sample_codec=audio_codec;
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mdp.vompevent.audio_sample_bytes=audio_length;
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mdp.vompevent.audio_sample_starttime=start_time;
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mdp.vompevent.audio_sample_endtime=end_time;
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}
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int i;
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long long now=overlay_gettime_ms();
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for(i=0;i<vomp_interested_usock_count;i++)
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if (vomp_interested_expiries[i]>=now) {
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overlay_mdp_reply(mdp_named.poll.fd,
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vomp_interested_usocks[i],
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vomp_interested_usock_lengths[i],
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&mdp);
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}
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return 0;
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}
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int vomp_send_mdp_status(vomp_call_state *call){
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return vomp_send_mdp_status_audio(call,0,0,0,NULL,0);
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}
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// send call state updates if required.
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int vomp_update(vomp_call_state *call){
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int combined_status=(call->remote.state<<4)|call->local.state;
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if (call->last_sent_status==combined_status)
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return 0;
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DEBUGF("Call state changed to %d %d, sending updates",call->local.state, call->remote.state);
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call->last_sent_status=combined_status;
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// tell the remote party
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vomp_send_status_remote(call);
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// tell monitor clients
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if (monitor_socket_count)
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monitor_call_status(call);
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// tell mdp clients
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if (vomp_interested_usock_count)
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vomp_send_mdp_status(call);
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return 0;
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}
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int vomp_call_start_audio(vomp_call_state *call)
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{
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call->audio_started=1;
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return WHY("Not implemented");
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}
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// check a small circular buffer of recently seen audio
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// we're not trying to be perfect here, we still expect all clients to reorder and filter duplicates
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int vomp_audio_already_seen(vomp_call_state *call, unsigned int end_time){
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int i;
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for(i=0;i<VOMP_MAX_RECENT_SAMPLES *4;i++)
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if (call->seen_samples[i]==end_time)
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return 1;
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call->seen_samples[call->sample_pos]=end_time;
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call->sample_pos++;
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if (call->sample_pos>=VOMP_MAX_RECENT_SAMPLES *4)
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call->sample_pos=0;
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return 0;
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}
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int vomp_process_audio(vomp_call_state *call,unsigned int sender_duration,overlay_mdp_frame *mdp)
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{
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int ofs=14;
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// if (mdp->in.payload_length>14)
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// WHYF("got here (payload has %d bytes)",mdp->in.payload_length);
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/* Get end time marker for sample block collection */
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unsigned int e=0, s=0, duration;
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e=mdp->in.payload[ofs++]<<24;
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e|=mdp->in.payload[ofs++]<<16;
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e|=mdp->in.payload[ofs++]<<8;
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e|=mdp->in.payload[ofs++]<<0;
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sender_duration = (e&0xFFFF0000)|sender_duration;
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if (0)
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DEBUGF("Jitter %d, %d",sender_duration -e,(overlay_gettime_ms()-call->create_time)-e);
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while(ofs<mdp->in.payload_length)
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{
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int codec=mdp->in.payload[ofs];
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// WHYF("Spotted a %s sample block",vomp_describe_codec(codec));
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if (!codec||vomp_sample_size(codec)<0) break;
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if ((ofs+1+vomp_sample_size(codec))>mdp->in.payload_length) break;
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/* work out start-time from end-time less duration of included sample(s).
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XXX - Assumes only non-adaptive codecs. */
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s = e-vomp_codec_timespan(codec)+1;
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/* Pass audio frame to all registered listeners */
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if (!vomp_audio_already_seen(call, e)){
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if (vomp_interested_usock_count)
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vomp_send_mdp_status_audio(call, codec, s, e,
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&mdp->in.payload[ofs+1],
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vomp_sample_size(codec)
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);
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if (monitor_socket_count)
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monitor_send_audio(call, codec, s, e,
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&mdp->in.payload[ofs+1],
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vomp_sample_size(codec)
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);
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}
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ofs+=1+vomp_sample_size(codec);
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e=s-1;
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}
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return 0;
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}
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int vomp_call_stop_audio(vomp_call_state *call)
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{
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call->audio_started=0;
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return WHY("Not implemented");
|
|
}
|
|
|
|
int vomp_call_start_ringing(vomp_call_state *call)
|
|
{
|
|
/* We just need to set the flag to say that we are ringing.
|
|
Interested listeners and far end will be informed via vomp_send_status() */
|
|
call->ringing=1;
|
|
fprintf(stderr,"RING RING!\n");
|
|
return 0;
|
|
}
|
|
|
|
int vomp_call_destroy(vomp_call_state *call)
|
|
{
|
|
/* do some general clean ups */
|
|
if (call->audio_started) vomp_call_stop_audio(call);
|
|
if (call->ringing) call->ringing=0;
|
|
|
|
fprintf(stderr,"Destroying call %s <--> %s\n",
|
|
call->local.did,call->remote.did);
|
|
|
|
/* tell everyone the call has died */
|
|
call->local.state=VOMP_STATE_CALLENDED; call->remote.state=VOMP_STATE_CALLENDED;
|
|
|
|
vomp_update(call);
|
|
|
|
/* now release the call structure */
|
|
int i = (call - vomp_call_states);
|
|
unschedule(&call->alarm);
|
|
|
|
vomp_call_count--;
|
|
if (i!=vomp_call_count){
|
|
unschedule(&vomp_call_states[vomp_call_count].alarm);
|
|
bcopy(&vomp_call_states[vomp_call_count],
|
|
call,
|
|
sizeof(vomp_call_state));
|
|
schedule(&call->alarm);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int vomp_dial(unsigned char *local_sid, unsigned char *remote_sid, char *local_did, char *remote_did){
|
|
/* TODO use local_did and remote_did start putting the call together.
|
|
These need to be passed to the node being called to provide caller id,
|
|
and potentially handle call-routing, e.g., if it is a gateway.
|
|
*/
|
|
DEBUG("Dialing\n");
|
|
|
|
if (vomp_call_count>=VOMP_MAX_CALLS)
|
|
return WHY("All call slots in use");
|
|
|
|
/* allocate unique call session token, which is how the client will
|
|
refer to this call during its life */
|
|
vomp_call_state *call=vomp_create_call(
|
|
remote_sid,
|
|
local_sid,
|
|
0,
|
|
0,
|
|
VOMP_STATE_NOCALL,
|
|
VOMP_STATE_CALLPREP
|
|
);
|
|
|
|
// remember that we initiated this call, not the other party
|
|
call->initiated_call = 1;
|
|
|
|
/* send status update to remote, thus causing call to be created
|
|
(hopefully) at far end. */
|
|
vomp_update(call);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int vomp_pickup(vomp_call_state *call){
|
|
if (call){
|
|
WHY("Picking up");
|
|
if (call->local.state!=VOMP_STATE_RINGINGIN)
|
|
return WHY("Call is not ringing");
|
|
|
|
call->local.state=VOMP_STATE_INCALL;
|
|
call->create_time=overlay_gettime_ms();
|
|
call->ringing=0;
|
|
/* state machine does job of starting audio stream, just tell everyone about
|
|
the changed state. */
|
|
vomp_update(call);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int vomp_hangup(vomp_call_state *call){
|
|
if (call){
|
|
WHY("Hanging up");
|
|
if (call->local.state==VOMP_STATE_INCALL) vomp_call_stop_audio(call);
|
|
call->local.state=VOMP_STATE_CALLENDED;
|
|
vomp_update(call);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/* An MDP message of type MDP_VOMPEVENT received from the unix domain socket.
|
|
This is how user tasks request telephone calls and receive updated status
|
|
and audio from the call. We need the receiver socket so that we can
|
|
route the events back to wherever they should be going.
|
|
XXX - We should have some means of authenticating/protecting this interface
|
|
so that any old process cannot request a mesh call. Although, in fairness,
|
|
the user will know about the call because the call display will come up.
|
|
*/
|
|
|
|
int vomp_mdp_event(overlay_mdp_frame *mdp,
|
|
struct sockaddr_un *recvaddr,int recvaddrlen)
|
|
{
|
|
/* Frames from the user can take only a few forms:
|
|
- announce interest in call state.
|
|
- withdraw interest in call state.
|
|
- place a call (SID+DID combination)
|
|
- deliver audio for sending
|
|
- indicate pickup, hangup or call reject
|
|
|
|
We then send back all sorts of relevant call state information as well as
|
|
transported audio. In particular we inform when the call state changes,
|
|
including if any error has occurred.
|
|
*/
|
|
if (0)
|
|
DEBUGF("Flags=0x%x\n",mdp->vompevent.flags);
|
|
|
|
switch(mdp->vompevent.flags)
|
|
{
|
|
case VOMPEVENT_REGISTERINTEREST:
|
|
WHY("Request to register interest");
|
|
/* put unix domain socket on record to send call state event and audio to. */
|
|
{
|
|
int i;
|
|
int candidate=-1;
|
|
long long now=overlay_gettime_ms();
|
|
for(i=0;i<vomp_interested_usock_count;i++)
|
|
{
|
|
if (vomp_interested_usock_lengths[i]==recvaddrlen)
|
|
if (!memcmp(recvaddr->sun_path,
|
|
vomp_interested_usocks[i],recvaddrlen))
|
|
/* found it -- so we are already monitoring this one */
|
|
return overlay_mdp_reply_error(mdp_named.poll.fd,recvaddr,recvaddrlen,
|
|
0,"Success");
|
|
if (vomp_interested_expiries[i]<now) candidate=i;
|
|
}
|
|
if (i>=vomp_interested_usock_count&&(candidate>-1)) i=candidate;
|
|
/* not currently on the list, so add */
|
|
if (i<VOMP_MAX_INTERESTED) {
|
|
if (vomp_interested_usocks[i]) {
|
|
free(vomp_interested_usocks[i]);
|
|
vomp_interested_usocks[i]=NULL;
|
|
}
|
|
vomp_interested_usocks[i]=malloc(recvaddrlen);
|
|
if (!vomp_interested_usocks[i])
|
|
return overlay_mdp_reply_error(mdp_named.poll.fd, recvaddr,recvaddrlen,
|
|
4002,"Out of memory");
|
|
bcopy(recvaddr,vomp_interested_usocks[i],
|
|
recvaddrlen);
|
|
vomp_interested_usock_lengths[i]=recvaddrlen;
|
|
vomp_interested_expiries[i]=overlay_gettime_ms()+60000;
|
|
if (i==vomp_interested_usock_count) vomp_interested_usock_count++;
|
|
|
|
if (mdp->vompevent.supported_codecs[0]) {
|
|
/* Replace set of locally supported codecs */
|
|
for(i=0;i<256;i++) vomp_local_codec_list[i]=0;
|
|
for(i=0;(i<256)&&mdp->vompevent.supported_codecs[i];i++)
|
|
{
|
|
vomp_local_codec_list[mdp->vompevent.supported_codecs[i]]=1;
|
|
}
|
|
}
|
|
|
|
return overlay_mdp_reply_error
|
|
(mdp_named.poll.fd,recvaddr,recvaddrlen,0,"Success");
|
|
} else {
|
|
return overlay_mdp_reply_error
|
|
(mdp_named.poll.fd,recvaddr,recvaddrlen,
|
|
4003,"Too many listeners (try again in a minute?)");
|
|
}
|
|
}
|
|
break;
|
|
case VOMPEVENT_WITHDRAWINTEREST:
|
|
/* opposite of above */
|
|
WHY("Request to withdraw interest");
|
|
{
|
|
int i;
|
|
for(i=0;i<vomp_interested_usock_count;i++)
|
|
{
|
|
if (vomp_interested_usock_lengths[i]==recvaddrlen)
|
|
if (!memcmp(recvaddr->sun_path,
|
|
vomp_interested_usocks[i],recvaddrlen))
|
|
{
|
|
/* found it -- so we are already monitoring this one */
|
|
free(vomp_interested_usocks[i]);
|
|
if (i<vomp_interested_usock_count-1)
|
|
{
|
|
int swap=vomp_interested_usock_count-1;
|
|
vomp_interested_usock_lengths[i]
|
|
=vomp_interested_usock_lengths[swap];
|
|
vomp_interested_usocks[i]=vomp_interested_usocks[swap];
|
|
vomp_interested_expiries[i]=vomp_interested_expiries[swap];
|
|
}
|
|
vomp_interested_usock_count--;
|
|
return overlay_mdp_reply_error
|
|
(mdp_named.poll.fd,recvaddr,recvaddrlen,
|
|
0,"Success. You have been removed.");
|
|
}
|
|
}
|
|
return overlay_mdp_reply_error
|
|
(mdp_named.poll.fd,recvaddr,recvaddrlen,
|
|
0,"Success. You were never listening.");
|
|
}
|
|
break;
|
|
case VOMPEVENT_CALLINFO:
|
|
{
|
|
/* provide call endpoint info to user */
|
|
vomp_call_state *call
|
|
=vomp_find_call_by_session(mdp->vompevent.call_session_token);
|
|
|
|
/* collect call info and send to requestor */
|
|
overlay_mdp_frame mdpreply;
|
|
bzero(&mdpreply,sizeof(mdpreply));
|
|
mdpreply.packetTypeAndFlags=MDP_VOMPEVENT;
|
|
mdpreply.vompevent.flags=VOMPEVENT_CALLINFO;
|
|
mdpreply.vompevent.call_session_token=mdp->vompevent.call_session_token;
|
|
if (call) {
|
|
if (call->ringing) mdpreply.vompevent.flags|=VOMPEVENT_RINGING;
|
|
if (call->audio_started)
|
|
mdpreply.vompevent.flags|=VOMPEVENT_AUDIOSTREAMING;
|
|
if (call->remote.state==VOMP_STATE_CALLENDED)
|
|
mdpreply.vompevent.flags|=VOMPEVENT_CALLENDED;
|
|
bcopy(call->local.sid,mdpreply.vompevent.local_sid,SID_SIZE);
|
|
bcopy(call->remote.sid,mdpreply.vompevent.remote_sid,SID_SIZE);
|
|
bcopy(call->local.did,mdpreply.vompevent.local_did,64);
|
|
bcopy(call->remote.did,mdpreply.vompevent.remote_did,64);
|
|
dump_vomp_status();
|
|
} else
|
|
if (mdp->vompevent.call_session_token)
|
|
/* let the requestor know that the requested call doesn't exist */
|
|
mdpreply.vompevent.flags|=VOMPEVENT_ERROR;
|
|
|
|
/* and provide a quick summary of all calls in progress */
|
|
int i;
|
|
for(i=0;i<vomp_call_count;i++)
|
|
{
|
|
mdpreply.vompevent.other_calls_sessions[i]
|
|
=vomp_call_states[i].local.session;
|
|
mdpreply.vompevent.other_calls_states[i]
|
|
=vomp_call_states[i].local.state;
|
|
}
|
|
|
|
return overlay_mdp_reply(mdp_named.poll.fd,recvaddr,recvaddrlen,&mdpreply);
|
|
}
|
|
break;
|
|
case VOMPEVENT_DIAL:
|
|
if (vomp_dial(
|
|
mdp->vompevent.local_sid,
|
|
mdp->vompevent.remote_sid,
|
|
NULL,
|
|
NULL))
|
|
return overlay_mdp_reply_error
|
|
(mdp_named.poll.fd,recvaddr,recvaddrlen,4004,
|
|
"Unable to place call");
|
|
else{
|
|
int result= overlay_mdp_reply_error
|
|
(mdp_named.poll.fd,recvaddr,recvaddrlen,0, "Success");
|
|
if (result) WHY("Failed to send MDP reply");
|
|
return result;
|
|
}
|
|
break;
|
|
case VOMPEVENT_CALLREJECT: /* hangup is the same */
|
|
{
|
|
vomp_call_state *call
|
|
=vomp_find_call_by_session(mdp->vompevent.call_session_token);
|
|
if (!call)
|
|
return overlay_mdp_reply_error
|
|
(mdp_named.poll.fd,recvaddr,recvaddrlen,4006,
|
|
"No such call");
|
|
|
|
vomp_hangup(call);
|
|
|
|
return overlay_mdp_reply_error(mdp_named.poll.fd,
|
|
recvaddr,recvaddrlen,0,"Success");
|
|
}
|
|
break;
|
|
case VOMPEVENT_PICKUP:
|
|
{
|
|
vomp_call_state *call
|
|
=vomp_find_call_by_session(mdp->vompevent.call_session_token);
|
|
if (!call)
|
|
return overlay_mdp_reply_error
|
|
(mdp_named.poll.fd,recvaddr,recvaddrlen,4006,
|
|
"No such call");
|
|
|
|
if (vomp_pickup(call))
|
|
return overlay_mdp_reply_error(mdp_named.poll.fd,
|
|
recvaddr,recvaddrlen,4009,
|
|
"Call is not RINGINGIN, so cannot be picked up");
|
|
else
|
|
return overlay_mdp_reply_error(mdp_named.poll.fd,
|
|
recvaddr,recvaddrlen,0,"Success");
|
|
}
|
|
break;
|
|
case VOMPEVENT_AUDIOPACKET: /* user supplying audio */
|
|
{
|
|
// WHY("Audio packet arrived");
|
|
vomp_call_state *call
|
|
=vomp_find_call_by_session(mdp->vompevent.call_session_token);
|
|
if (call) {
|
|
return vomp_send_status_remote_audio(call,
|
|
mdp->vompevent.audio_sample_codec,
|
|
&mdp->vompevent.audio_bytes[0],
|
|
vomp_sample_size(mdp->vompevent.audio_sample_codec));
|
|
}
|
|
else WHY("audio packet had invalid call session token");
|
|
}
|
|
break;
|
|
default:
|
|
/* didn't understand it, so respond with an error */
|
|
return overlay_mdp_reply_error(mdp_named.poll.fd,
|
|
recvaddr,recvaddrlen,4001,
|
|
"Invalid VOMPEVENT request (use DIAL,HANGUP,CALLREJECT,AUDIOSTREAMING,REGISTERINTERST,WITHDRAWINTERST only)");
|
|
|
|
}
|
|
|
|
return WHY("Not implemented");
|
|
}
|
|
|
|
int vomp_extract_remote_codec_list(vomp_call_state *call,overlay_mdp_frame *mdp)
|
|
{
|
|
int i;
|
|
if (0) {
|
|
WHY("Receiving list of remote codecs");
|
|
dump("codec list mdp frame",
|
|
(unsigned char *)&mdp->in.payload[0],mdp->in.payload_length);
|
|
}
|
|
for(i=0;mdp->in.payload[14+i]&&(i<256)
|
|
&&((14+i)<mdp->in.payload_length);i++)
|
|
{
|
|
if (0) WHYF("populating remote codec list with %s",
|
|
vomp_describe_codec(mdp->in.payload[14+i]));
|
|
call->remote_codec_list[mdp->in.payload[14+i]]=1;
|
|
}
|
|
// TODO send codec list to monitor clients
|
|
return 0;
|
|
}
|
|
|
|
/*
|
|
Simplified call state lifecycle;
|
|
|
|
(we dial)
|
|
> CALLPREP + codecs
|
|
< NOCALL + codecs
|
|
> RINGOUT
|
|
< RINGIN (they start ringing)
|
|
(we start "they are ringing" tone)
|
|
(they pickup)
|
|
< INCALL
|
|
> INCALL
|
|
(either hangup, or state error)
|
|
> CALLENDED
|
|
< CALLENDED
|
|
*/
|
|
|
|
/* At this point we know the MDP frame is addressed to the VoMP port, but
|
|
we have not inspected the contents. As these frames are wire-format, we
|
|
must pay attention to endianness. */
|
|
int vomp_mdp_received(overlay_mdp_frame *mdp)
|
|
{
|
|
if (mdp->packetTypeAndFlags&(MDP_NOCRYPT|MDP_NOSIGN))
|
|
{
|
|
/* stream-crypted audio frame */
|
|
return WHY("not implemented");
|
|
}
|
|
|
|
/* only auth-crypted frames make it this far */
|
|
|
|
vomp_call_state *call=NULL;
|
|
|
|
switch(mdp->in.payload[0]) {
|
|
case 0x01: /* Ordinary VoMP state+optional audio frame */
|
|
{
|
|
int recvr_state=mdp->in.payload[1]>>4;
|
|
int sender_state=mdp->in.payload[1]&0xf;
|
|
unsigned int recvr_session=
|
|
(mdp->in.payload[8]<<16)|(mdp->in.payload[9]<<8)|mdp->in.payload[10];
|
|
unsigned int sender_session=
|
|
(mdp->in.payload[11]<<16)|(mdp->in.payload[12]<<8)|mdp->in.payload[13];
|
|
int sender_seq=(mdp->in.payload[4]<<8)+mdp->in.payload[5];
|
|
|
|
// cyclic ~1 minute timer...
|
|
unsigned int sender_duration = (mdp->in.payload[6]<<8) | mdp->in.payload[7];
|
|
|
|
/* wants to create a call session.
|
|
Main aim here: replay protection. An adversary should not be able to
|
|
replay previous VoMP packets to cause any action. We do this by
|
|
allocating a new session number for each call. As an adversary may be
|
|
trying to use such replays to cause a denial of service attack we need
|
|
to be able to track multiple potential session numbers even from the
|
|
same SID. */
|
|
|
|
call=vomp_find_or_create_call(mdp->in.src.sid,mdp->in.dst.sid,
|
|
sender_session,recvr_session,
|
|
sender_state,recvr_state);
|
|
|
|
if (!call)
|
|
return WHY("Unable to find or create call");
|
|
|
|
if (!recvr_session)
|
|
WHY("recvr_session==0, created call.");
|
|
|
|
|
|
// TODO ignore state changes if sequence is stale?
|
|
// TODO ignore state changes that seem to go backwards?
|
|
|
|
if ((!vomp_interested_usock_count)
|
|
&&(!monitor_socket_count))
|
|
{
|
|
/* No registered listener, so we cannot answer the call, so just reject
|
|
it. */
|
|
if (0) WHYF("Rejecting call due to lack of a listener: states=%d,%d",
|
|
call->local.state,sender_state);
|
|
|
|
call->local.state=VOMP_STATE_CALLENDED;
|
|
/* now let the state machine progress to destroy the call */
|
|
}
|
|
|
|
if (call->local.state < VOMP_STATE_RINGINGOUT && sender_state < VOMP_STATE_RINGINGOUT){
|
|
// the other party should have given us their list of supported codecs
|
|
vomp_extract_remote_codec_list(call,mdp);
|
|
}
|
|
|
|
if (sender_state==VOMP_STATE_CALLENDED){
|
|
/* For whatever reason, the far end has given up on the call,
|
|
so we must also move to CALLENDED no matter what state we were in */
|
|
|
|
if (call->audio_started) vomp_call_stop_audio(call);
|
|
if (call->ringing) call->ringing=0;
|
|
call->local.state=VOMP_STATE_CALLENDED;
|
|
}
|
|
|
|
/* Consider states: our actual state, sender state, what the sender thinks
|
|
our state is, and what we think the sender's state is. But largely it
|
|
breaks down to what we think our state is, and what they think their
|
|
state is. That leaves us with just 6X6=36 cases.
|
|
*/
|
|
int combined_state=call->local.state<<3 | sender_state;
|
|
|
|
switch(combined_state) {
|
|
case (VOMP_STATE_NOCALL<<3)|VOMP_STATE_CALLPREP:
|
|
/* The remote party is in the call-prep state tryng to dial us.
|
|
We'll send them our codec list, then they can tell us to ring.
|
|
*/
|
|
break;
|
|
|
|
case (VOMP_STATE_RINGINGIN<<3)|VOMP_STATE_RINGINGOUT:
|
|
/* they are ringing us and we are ringing. Lets keep doing that. */
|
|
case (VOMP_STATE_NOCALL<<3)|VOMP_STATE_RINGINGOUT:
|
|
/* We have have issued a session, the remote party is now indicating
|
|
that they would like us to start ringing.
|
|
So change our state to RINGINGIN. */
|
|
|
|
if (call->initiated_call)
|
|
// hey, quit it, we were trying to call you.
|
|
call->local.state=VOMP_STATE_CALLENDED;
|
|
else{
|
|
// TODO fail the call if we can't find a codec you know
|
|
call->local.state=VOMP_STATE_RINGINGIN;
|
|
if (!call->ringing) vomp_call_start_ringing(call);
|
|
}
|
|
break;
|
|
|
|
case (VOMP_STATE_CALLPREP<<3)|VOMP_STATE_NOCALL:
|
|
case (VOMP_STATE_CALLPREP<<3)|VOMP_STATE_CALLPREP:
|
|
/* We are getting ready to ring, and the other end has issued a session
|
|
number, (and may be calling us at the same time).
|
|
Now is the time to ring out.
|
|
However, until the remote party has acknowledged with RINGIN,
|
|
don't indicate their ringing state to the user.
|
|
*/
|
|
if (call->initiated_call){
|
|
// TODO fail the call if we can't agree on codec's
|
|
call->local.state=VOMP_STATE_RINGINGOUT;
|
|
}else{
|
|
call->local.state=VOMP_STATE_CALLENDED;
|
|
}
|
|
break;
|
|
|
|
case (VOMP_STATE_RINGINGOUT<<3)|VOMP_STATE_NOCALL:
|
|
case (VOMP_STATE_RINGINGOUT<<3)|VOMP_STATE_CALLPREP:
|
|
/* We are calling them, and they have not yet answered, just wait */
|
|
break;
|
|
|
|
case (VOMP_STATE_RINGINGOUT<<3)|VOMP_STATE_RINGINGIN:
|
|
/* we are calling them and they have acknowledged it.
|
|
Now we can play a tone to indicate they are ringing */
|
|
break;
|
|
|
|
case (VOMP_STATE_RINGINGOUT<<3)|VOMP_STATE_RINGINGOUT:
|
|
/* Woah, we're trying to dial each other?? That must have been well timed.
|
|
Jump to INCALL and start audio */
|
|
call->local.state=VOMP_STATE_INCALL;
|
|
// reset create time when call is established
|
|
call->create_time=overlay_gettime_ms();
|
|
break;
|
|
|
|
case (VOMP_STATE_INCALL<<3)|VOMP_STATE_RINGINGOUT:
|
|
/* we think the call is in progress, but the far end hasn't replied yet
|
|
Just wait. */
|
|
break;
|
|
|
|
case (VOMP_STATE_RINGINGOUT<<3)|VOMP_STATE_INCALL:
|
|
/* They have answered, we can jump to incall as well */
|
|
call->local.state=VOMP_STATE_INCALL;
|
|
// reset create time when call is established
|
|
call->create_time=overlay_gettime_ms();
|
|
call->ringing=0;
|
|
// Fall through
|
|
case (VOMP_STATE_INCALL<<3)|VOMP_STATE_INCALL:
|
|
/* play any audio that they have sent us. */
|
|
if (!call->audio_started) {
|
|
if (vomp_call_start_audio(call)) call->local.codec=VOMP_CODEC_ENGAGED;
|
|
}
|
|
vomp_process_audio(call,sender_duration,mdp);
|
|
break;
|
|
|
|
case (VOMP_STATE_CALLENDED<<3)|VOMP_STATE_NOCALL:
|
|
case (VOMP_STATE_CALLENDED<<3)|VOMP_STATE_CALLPREP:
|
|
case (VOMP_STATE_CALLENDED<<3)|VOMP_STATE_RINGINGOUT:
|
|
case (VOMP_STATE_CALLENDED<<3)|VOMP_STATE_RINGINGIN:
|
|
case (VOMP_STATE_CALLENDED<<3)|VOMP_STATE_INCALL:
|
|
case (VOMP_STATE_CALLENDED<<3)|VOMP_STATE_CALLENDED:
|
|
/* If we ended the call, we'll wait for the far end to reply before destroying it */
|
|
break;
|
|
|
|
default:
|
|
/*
|
|
Any state not explicitly listed above is considered invalid and possibly stale,
|
|
the packet will be completely ignored.
|
|
*/
|
|
return 0;
|
|
}
|
|
|
|
call->remote.sequence=sender_seq;
|
|
call->remote.state=sender_state;
|
|
call->last_activity=overlay_gettime_ms();
|
|
|
|
// TODO if we hear a stale echo of our state should we force another outgoing packet now?
|
|
// will that always cause 2 outgoing packets?
|
|
|
|
/* send an update to the call status if required */
|
|
vomp_update(call);
|
|
|
|
if (call->remote.state==VOMP_STATE_CALLENDED
|
|
&&call->local.state==VOMP_STATE_CALLENDED)
|
|
return vomp_call_destroy(call);
|
|
}
|
|
return 0;
|
|
break;
|
|
default:
|
|
/* unsupported VoMP frame */
|
|
WHYF("Unsupported VoMP frame type = 0x%02x",mdp->in.payload[0]);
|
|
break;
|
|
}
|
|
|
|
return WHY("Malformed VoMP MDP packet?");
|
|
}
|
|
|
|
char *vomp_describe_state(int state)
|
|
{
|
|
switch(state) {
|
|
case VOMP_STATE_CALLENDED: return "CALLENDED";
|
|
case VOMP_STATE_INCALL: return "INCALL";
|
|
case VOMP_STATE_RINGINGIN: return "RINGINGIN";
|
|
case VOMP_STATE_RINGINGOUT: return "RINGINGOUT";
|
|
case VOMP_STATE_CALLPREP: return "CALLPREP";
|
|
case VOMP_STATE_NOCALL: return "NOCALL";
|
|
}
|
|
return "UNKNOWN";
|
|
}
|
|
|
|
int dump_vomp_status()
|
|
{
|
|
int i;
|
|
printf(">>> Active VoMP call states:\n");
|
|
for(i=0;i<vomp_call_count;i++)
|
|
{
|
|
printf("%s/%06x\n-> %s/%06x\n (%s -> %s)\n",
|
|
alloca_tohex_sid(vomp_call_states[i].local.sid),
|
|
vomp_call_states[i].local.session,
|
|
alloca_tohex_sid(vomp_call_states[i].remote.sid),
|
|
vomp_call_states[i].remote.session,
|
|
vomp_call_states[i].local.did,
|
|
vomp_call_states[i].remote.did);
|
|
printf(" local state=%s, remote state=%s\n",
|
|
vomp_describe_state(vomp_call_states[i].local.state),
|
|
vomp_describe_state(vomp_call_states[i].remote.state));
|
|
}
|
|
if (!vomp_call_count) printf("No active calls\n");
|
|
return 0;
|
|
}
|
|
|
|
char *vomp_describe_codec(int c)
|
|
{
|
|
switch(c) {
|
|
case VOMP_CODEC_NONE: return "none";
|
|
case VOMP_CODEC_CODEC2_2400: return "CODEC2@1400";
|
|
case VOMP_CODEC_CODEC2_1400: return "CODEC2@2400";
|
|
case VOMP_CODEC_GSMHALF: return "GSM-half-rate";
|
|
case VOMP_CODEC_GSMFULL: return "GSM-full-rate";
|
|
case VOMP_CODEC_16SIGNED: return "16bit-raw";
|
|
case VOMP_CODEC_8ULAW: return "8bit-uLaw";
|
|
case VOMP_CODEC_8ALAW: return "8bit-aLaw";
|
|
case VOMP_CODEC_PCM: return "PCM@8KHz";
|
|
case VOMP_CODEC_DTMF: return "DTMF";
|
|
case VOMP_CODEC_ENGAGED: return "Engaged-tone";
|
|
case VOMP_CODEC_ONHOLD: return "On-Hold";
|
|
case VOMP_CODEC_CALLERID: return "CallerID";
|
|
}
|
|
return "unknown";
|
|
}
|
|
|
|
int vomp_sample_size(int c)
|
|
{
|
|
switch(c) {
|
|
case VOMP_CODEC_NONE: return 0;
|
|
case VOMP_CODEC_CODEC2_2400: return 7; /* actually 2550bps, 51 bits per 20ms,
|
|
but using whole byte here, so 2800bps */
|
|
case VOMP_CODEC_CODEC2_1400: return 7; /* per 40ms */
|
|
case VOMP_CODEC_GSMHALF: return 14; /* check. 5.6kbits */
|
|
case VOMP_CODEC_GSMFULL: return 33; /* padded to 13.2kbit/sec */
|
|
case VOMP_CODEC_16SIGNED: return 320; /* 8000x2bytes*0.02sec */
|
|
case VOMP_CODEC_8ULAW: return 160;
|
|
case VOMP_CODEC_8ALAW: return 160;
|
|
case VOMP_CODEC_PCM: return 320;
|
|
case VOMP_CODEC_DTMF: return 1;
|
|
case VOMP_CODEC_ENGAGED: return 0;
|
|
case VOMP_CODEC_ONHOLD: return 0;
|
|
case VOMP_CODEC_CALLERID: return 32;
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
int vomp_codec_timespan(int c)
|
|
{
|
|
switch(c) {
|
|
case VOMP_CODEC_NONE: return 1;
|
|
case VOMP_CODEC_CODEC2_2400: return 20;
|
|
case VOMP_CODEC_CODEC2_1400: return 40;
|
|
case VOMP_CODEC_GSMHALF: return 20;
|
|
case VOMP_CODEC_GSMFULL: return 20;
|
|
case VOMP_CODEC_16SIGNED: return 20;
|
|
case VOMP_CODEC_8ULAW: return 20;
|
|
case VOMP_CODEC_8ALAW: return 20;
|
|
case VOMP_CODEC_PCM: return 20;
|
|
case VOMP_CODEC_DTMF: return 80;
|
|
case VOMP_CODEC_ENGAGED: return 20;
|
|
case VOMP_CODEC_ONHOLD: return 20;
|
|
case VOMP_CODEC_CALLERID: return 0;
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
|
|
int app_vomp_status(int argc, const char *const *argv, struct command_line_option *o)
|
|
{
|
|
overlay_mdp_frame mdp;
|
|
bzero(&mdp,sizeof(mdp));
|
|
|
|
mdp.packetTypeAndFlags=MDP_VOMPEVENT;
|
|
mdp.vompevent.flags=VOMPEVENT_CALLINFO;
|
|
if (overlay_mdp_send(&mdp,MDP_AWAITREPLY,5000))
|
|
{
|
|
WHY("Current call state information request failed.");
|
|
if (mdp.packetTypeAndFlags==MDP_ERROR&&mdp.error.error)
|
|
fprintf(stderr,"MDP: error=%d, message='%s'\n",
|
|
mdp.error.error,mdp.error.message);
|
|
overlay_mdp_client_done();
|
|
return -1;
|
|
}
|
|
if (mdp.packetTypeAndFlags!=MDP_VOMPEVENT) {
|
|
WHYF("Received incorrect reply type from server (received MDP message type 0x%04x)\n",mdp.packetTypeAndFlags);
|
|
overlay_mdp_client_done();
|
|
return -1;
|
|
}
|
|
if (mdp.vompevent.flags!=VOMPEVENT_CALLINFO) {
|
|
WHYF("Received incorrect reply type from server (received VoMP message type 0x%04x)\n",mdp.vompevent.flags);
|
|
overlay_mdp_client_done();
|
|
return -1;
|
|
}
|
|
int i;
|
|
int count=0;
|
|
overlay_mdp_frame mdp2;
|
|
bzero(&mdp2,sizeof(mdp2));
|
|
for(i=0;i<VOMP_MAX_CALLS;i++)
|
|
if (mdp.vompevent.other_calls_sessions[i])
|
|
{
|
|
count++;
|
|
fprintf(stderr,"%06x:%s:",
|
|
mdp.vompevent.other_calls_sessions[i],
|
|
vomp_describe_state(mdp.vompevent.other_calls_states[i]));
|
|
mdp2.packetTypeAndFlags=MDP_VOMPEVENT;
|
|
mdp2.vompevent.flags=VOMPEVENT_CALLINFO;
|
|
mdp2.vompevent.call_session_token=mdp.vompevent.other_calls_sessions[i];
|
|
if (overlay_mdp_send(&mdp2,MDP_AWAITREPLY,5000))
|
|
fprintf(stderr,"<server failed to provide detail>");
|
|
else
|
|
{
|
|
if (mdp2.vompevent.call_session_token!=mdp.vompevent.other_calls_sessions[i])
|
|
fprintf(stderr,"<strange reply from server (%04x, %04x, token %06x)>",
|
|
mdp.packetTypeAndFlags,mdp.vompevent.flags,
|
|
mdp2.vompevent.call_session_token);
|
|
else {
|
|
fprintf(stderr,"%s* -> %s* (%s -> %s)",
|
|
alloca_tohex(mdp2.vompevent.local_sid, 6),
|
|
alloca_tohex(mdp2.vompevent.remote_sid, 6),
|
|
strlen(mdp2.vompevent.local_did)
|
|
?mdp2.vompevent.local_did:"<no local number>",
|
|
strlen(mdp2.vompevent.remote_did)
|
|
?mdp2.vompevent.remote_did:"<no remote number>");
|
|
}
|
|
int i;
|
|
fprintf(stderr," supports");
|
|
for(i=0;i<256;i++)
|
|
if (mdp2.vompevent.supported_codecs[i])
|
|
fprintf(stderr," %s",vomp_describe_codec(i));
|
|
}
|
|
fprintf(stderr,"\n");
|
|
}
|
|
fprintf(stderr,"%d live call descriptors.\n",count);
|
|
return overlay_mdp_client_done();
|
|
}
|
|
|
|
int app_vomp_dial(int argc, const char *const *argv, struct command_line_option *o)
|
|
{
|
|
const char *sid,*did,*callerid;
|
|
cli_arg(argc, argv, o, "sid", &sid, NULL, "");
|
|
cli_arg(argc, argv, o, "did", &did, NULL, "");
|
|
cli_arg(argc, argv, o, "callerid", &callerid, NULL, NULL);
|
|
|
|
overlay_mdp_frame mdp;
|
|
bzero(&mdp,sizeof(mdp));
|
|
|
|
mdp.packetTypeAndFlags=MDP_VOMPEVENT;
|
|
mdp.vompevent.flags=VOMPEVENT_DIAL;
|
|
if (overlay_mdp_getmyaddr(0,&mdp.vompevent.local_sid[0])) return -1;
|
|
stowSid(&mdp.vompevent.remote_sid[0],0,sid);
|
|
printf("local_sid=%s\n",alloca_tohex_sid(mdp.vompevent.local_sid));
|
|
printf("remote_sid=%s from %s\n", alloca_tohex_sid(mdp.vompevent.remote_sid),sid);
|
|
|
|
if (overlay_mdp_send(&mdp,MDP_AWAITREPLY,5000))
|
|
{
|
|
WHY("Dial request failed.");
|
|
}
|
|
if (mdp.packetTypeAndFlags==MDP_ERROR&&mdp.error.error)
|
|
fprintf(stderr,"Dial request failed: error=%d, message='%s'\n",
|
|
mdp.error.error,mdp.error.message);
|
|
else
|
|
printf("Dial request accepted.\n");
|
|
|
|
return overlay_mdp_client_done();
|
|
}
|
|
|
|
int vomp_parse_dtmf_digit(char c)
|
|
{
|
|
if (c>='0'&&c<='9') return c-0x30;
|
|
switch (c) {
|
|
case 'a': case 'A': return 0xa;
|
|
case 'b': case 'B': return 0xb;
|
|
case 'c': case 'C': return 0xc;
|
|
case 'd': case 'D': return 0xd;
|
|
case '*': return 0xe;
|
|
case '#': return 0xf;
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
char vomp_dtmf_digit_to_char(int digit)
|
|
{
|
|
if (digit<0) return '?';
|
|
if (digit<10) return '0'+digit;
|
|
if (digit<0xe) return 'A'+digit-0xa;
|
|
if (digit==0xe) return '*';
|
|
if (digit==0xf) return '#';
|
|
return '?';
|
|
}
|
|
|
|
int app_vomp_dtmf(int argc, const char *const *argv, struct command_line_option *o)
|
|
{
|
|
const char *call_token;
|
|
const char *digits;
|
|
cli_arg(argc, argv, o, "call", &call_token, NULL, "");
|
|
cli_arg(argc, argv, o, "digits", &digits,NULL,"");
|
|
|
|
overlay_mdp_frame mdp;
|
|
bzero(&mdp,sizeof(mdp));
|
|
|
|
mdp.packetTypeAndFlags=MDP_VOMPEVENT;
|
|
mdp.vompevent.flags=VOMPEVENT_AUDIOPACKET;
|
|
mdp.vompevent.call_session_token=strtol(call_token,NULL,16);
|
|
|
|
/* One digit per sample block. */
|
|
mdp.vompevent.audio_sample_codec=VOMP_CODEC_DTMF;
|
|
mdp.vompevent.audio_sample_bytes=1;
|
|
|
|
int i;
|
|
for(i=0;i<strlen(digits);i++) {
|
|
int digit=vomp_parse_dtmf_digit(digits[i]);
|
|
if (digit<0) return WHYF("'%c' is not a DTMF digit.",digits[i]);
|
|
mdp.vompevent.audio_bytes[mdp.vompevent.audio_sample_bytes]
|
|
=(digit<<4); /* 80ms standard tone duration, so that it is a multiple
|
|
of the majority of codec time units (70ms is the nominal
|
|
DTMF tone length for most systems). */
|
|
if (overlay_mdp_send(&mdp,0,0)) WHY("Send DTMF failed.");
|
|
}
|
|
|
|
printf("DTMF digit(s) sent.\n");
|
|
|
|
return overlay_mdp_client_done();
|
|
}
|
|
|
|
|
|
int app_vomp_pickup(int argc, const char *const *argv, struct command_line_option *o)
|
|
{
|
|
const char *call_token;
|
|
cli_arg(argc, argv, o, "call", &call_token, NULL, "");
|
|
|
|
overlay_mdp_frame mdp;
|
|
bzero(&mdp,sizeof(mdp));
|
|
|
|
mdp.packetTypeAndFlags=MDP_VOMPEVENT;
|
|
mdp.vompevent.flags=VOMPEVENT_PICKUP;
|
|
mdp.vompevent.call_session_token=strtol(call_token,NULL,16);
|
|
|
|
if (overlay_mdp_send(&mdp,MDP_AWAITREPLY,5000))
|
|
{
|
|
WHY("Pickup request failed.");
|
|
}
|
|
if (mdp.packetTypeAndFlags==MDP_ERROR&&mdp.error.error)
|
|
fprintf(stderr,"Pickup request failed: error=%d, message='%s'\n",
|
|
mdp.error.error,mdp.error.message);
|
|
else
|
|
printf("Pickup request accepted.\n");
|
|
|
|
return overlay_mdp_client_done();
|
|
}
|
|
|
|
int app_vomp_hangup(int argc, const char *const *argv, struct command_line_option *o)
|
|
{
|
|
const char *call_token;
|
|
cli_arg(argc, argv, o, "call", &call_token, NULL, "");
|
|
|
|
overlay_mdp_frame mdp;
|
|
bzero(&mdp,sizeof(mdp));
|
|
|
|
mdp.packetTypeAndFlags=MDP_VOMPEVENT;
|
|
mdp.vompevent.flags=VOMPEVENT_HANGUP;
|
|
mdp.vompevent.call_session_token=strtol(call_token,NULL,16);
|
|
|
|
if (overlay_mdp_send(&mdp,MDP_AWAITREPLY,5000))
|
|
{
|
|
WHY("Hangup/reject request failed.");
|
|
}
|
|
if (mdp.packetTypeAndFlags==MDP_ERROR&&mdp.error.error)
|
|
fprintf(stderr,"Hangup/reject request failed: error=%d, message='%s'\n",
|
|
mdp.error.error,mdp.error.message);
|
|
else
|
|
printf("Hangup/reject request accepted.\n");
|
|
|
|
return overlay_mdp_client_done();
|
|
}
|
|
|
|
int app_vomp_monitor(int argc, const char *const *argv, struct command_line_option *o)
|
|
{
|
|
overlay_mdp_frame mdp;
|
|
bzero(&mdp,sizeof(mdp));
|
|
|
|
mdp.packetTypeAndFlags=MDP_VOMPEVENT;
|
|
mdp.vompevent.flags=VOMPEVENT_REGISTERINTEREST;
|
|
mdp.vompevent.supported_codecs[0]=VOMP_CODEC_DTMF;
|
|
mdp.vompevent.supported_codecs[1]=VOMP_CODEC_NONE;
|
|
|
|
if (overlay_mdp_send(&mdp,MDP_AWAITREPLY,5000))
|
|
{ WHY("Failed to register interest in telephony events.");
|
|
overlay_mdp_client_done();
|
|
if (mdp.packetTypeAndFlags==MDP_ERROR&&mdp.error.error)
|
|
fprintf(stderr," MDP Server error #%d: '%s'\n",
|
|
mdp.error.error,mdp.error.message);
|
|
return -1;
|
|
}
|
|
|
|
while(!servalShutdown) {
|
|
overlay_mdp_frame rx;
|
|
int ttl;
|
|
/* In theory we should be able to ask for a timeout of -1 for
|
|
infinity, but broken poll() and select() implementations on OSX
|
|
make this impossible. So one unnecessary check per second is
|
|
probably tolerable. */
|
|
if (overlay_mdp_client_poll(1000)>0)
|
|
if (!overlay_mdp_recv(&rx,&ttl))
|
|
{
|
|
switch(rx.packetTypeAndFlags) {
|
|
case MDP_ERROR:
|
|
fprintf(stderr,"MDP Server error #%d: '%s'\n",
|
|
rx.error.error,rx.error.message);
|
|
break;
|
|
case MDP_VOMPEVENT:
|
|
fprintf(stderr,"VoMP call descriptor %06x %s:%s",
|
|
rx.vompevent.call_session_token,
|
|
vomp_describe_state(rx.vompevent.local_state),
|
|
vomp_describe_state(rx.vompevent.remote_state));
|
|
if (rx.vompevent.flags&VOMPEVENT_RINGING)
|
|
fprintf(stderr," RINGING");
|
|
if (rx.vompevent.flags&VOMPEVENT_CALLENDED)
|
|
fprintf(stderr," CALLENDED");
|
|
if (rx.vompevent.flags&VOMPEVENT_CALLREJECT)
|
|
fprintf(stderr," CALLREJECTED");
|
|
if (rx.vompevent.flags&VOMPEVENT_CALLCREATED)
|
|
fprintf(stderr," CREATED");
|
|
if (rx.vompevent.flags&VOMPEVENT_AUDIOSTREAMING)
|
|
fprintf(stderr," AUDIOSTREAMING");
|
|
int i;
|
|
fprintf(stderr," codecs:");
|
|
for(i=0;i<256;i++)
|
|
if (rx.vompevent.supported_codecs[i])
|
|
fprintf(stderr," %s",vomp_describe_codec(i));
|
|
|
|
fprintf(stderr,"\n");
|
|
if (rx.vompevent.audio_sample_codec) {
|
|
fprintf(stderr," attached audio sample: codec=%s, len=%d\n",
|
|
vomp_describe_codec(rx.vompevent.audio_sample_codec),
|
|
rx.vompevent.audio_sample_bytes);
|
|
fprintf(stderr," sample covers %lldms - %lldms of call.\n",
|
|
rx.vompevent.audio_sample_starttime,
|
|
rx.vompevent.audio_sample_endtime);
|
|
}
|
|
break;
|
|
default:
|
|
fprintf(stderr,"Unknown message (type=0x%02x)\n",rx.packetTypeAndFlags);
|
|
}
|
|
}
|
|
|
|
}
|
|
|
|
mdp.packetTypeAndFlags=MDP_VOMPEVENT;
|
|
mdp.vompevent.flags=VOMPEVENT_WITHDRAWINTEREST;
|
|
if (overlay_mdp_send(&mdp,MDP_AWAITREPLY,5000))
|
|
{ WHY("Failed to deregister interest in telephony events.");
|
|
overlay_mdp_client_done(); return -1; }
|
|
if (mdp.packetTypeAndFlags==MDP_ERROR&&mdp.error.error) {
|
|
fprintf(stderr," MDP Server error #%d: '%s'\n",
|
|
mdp.error.error,mdp.error.message);
|
|
}
|
|
|
|
return overlay_mdp_client_done();
|
|
}
|
|
|
|
void vomp_process_tick(struct sched_ent *alarm){
|
|
char msg[32];
|
|
int len;
|
|
unsigned long long now = overlay_gettime_ms();
|
|
|
|
vomp_call_state *call = (vomp_call_state *)alarm;
|
|
|
|
/* See if any calls need to begin expiring
|
|
(current timeout is set at 2 minutes) */
|
|
if (call->local.state<VOMP_STATE_INCALL
|
|
&&((call->create_time+VOMP_CALL_TIMEOUT)<now))
|
|
{
|
|
/* timeout calls that haven't reached INCALL status, e.g.,
|
|
ringing. As well as sensible UX, it also prevents our call
|
|
slots getting full of cruft. */
|
|
vomp_call_destroy(call);
|
|
return;
|
|
} else if (call->last_activity+VOMP_CALL_TIMEOUT<now)
|
|
switch(call->local.state)
|
|
{
|
|
case VOMP_STATE_INCALL:
|
|
/* Timeout while call in progress, so end call.
|
|
Keep call structure hanging around for a bit so that we can
|
|
synchonrise with the far end if possible. */
|
|
call->local.state=VOMP_STATE_CALLENDED;
|
|
vomp_call_stop_audio(call);
|
|
call->last_activity=now;
|
|
break;
|
|
default:
|
|
/* Call timed out while not actually in progress, so just immmediately
|
|
tear the call down */
|
|
vomp_call_destroy(call);
|
|
return;
|
|
}
|
|
|
|
/* update everyone if the state has changed */
|
|
vomp_update(call);
|
|
/* force a packet to the other party. We are still here */
|
|
vomp_send_status_remote(call);
|
|
|
|
/* tell local monitor clients the call is still alive */
|
|
len = snprintf(msg,sizeof(msg) -1,"\nKEEPALIVE:%06x\n", call->local.session);
|
|
monitor_tell_clients(msg, len, MONITOR_VOMP);
|
|
|
|
alarm->alarm = overlay_gettime_ms() + VOMP_CALL_STATUS_INTERVAL;
|
|
schedule(alarm);
|
|
}
|