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1068 lines
34 KiB
C
1068 lines
34 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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Copyright (C) 2012 Serval Project Inc.
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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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#include "strbuf.h"
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#include "strlcpy.h"
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/*
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Typical call state lifecycle between 2 parties.
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Legend;
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# incoming command from monitor client
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$ outgoing monitor status
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<> vomp packet with state change sent across the network
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Monitor Init
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# MONITOR VOMP [supported codec list]
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Dialing
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// client requests an outgoing call
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# CALL [sid] [myDid] [TheirDid]
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> CALLPREP + codecs
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// let the client know what token we are going to use for the remainder of the call
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$ CALLTO [token] [mySid] [myDid] [TheirSid] [TheirDid]
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// allocate a session number and tell them our codecs,
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// but we don't need to do anything else yet,
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// this might be a replay attack
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< NOCALL + codecs
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// Ok, we have a network path, lets try to establish the call
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> RINGOUT
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// (Note that if both parties are trying to dial each other,
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// the call should jump straight to INCALL)
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// inform client about the call request
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$ CALLFROM [token] [mySid] [myDid] [TheirSid] [TheirDid]
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// Note that we may need to wait for other external processes
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// before a phone is actually ringing
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# RING [token]
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< RINGIN
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// All good, there's a phone out there ringing, you can indicate that to the user
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$ RINGING [token]
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Answering
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# PICKUP [token]
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< INCALL
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// The client can now start sending audio
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> INCALL
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$ INCALL [token]
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// The client can now start sending audio
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$ INCALL [token]
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Tell any clients that the call hasn't timed out yet
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(if servald is behaving this should be redundant, if it isn't behaving how do we hangup?)
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$ KEEPALIVE [token]
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Hanging up (may also be triggered on network or call establishment timeout)
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# HANGUP [token]
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> CALLENDED
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$ HANGUP [token]
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< CALLENDED
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$ HANGUP [token]
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*/
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// ideally these id's should only be used on the network, with monitor events to inform clients of state changes
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#define VOMP_STATE_NOCALL 1
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#define VOMP_STATE_CALLPREP 2
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#define VOMP_STATE_RINGINGOUT 3
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#define VOMP_STATE_RINGINGIN 4
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#define VOMP_STATE_INCALL 5
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#define VOMP_STATE_CALLENDED 6
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struct vomp_call_half {
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unsigned char sid[SID_SIZE];
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char did[64];
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unsigned char state;
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unsigned char codec;
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unsigned int session;
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unsigned int sequence;
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/* the following is from call creation, not start of audio flow */
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unsigned long long milliseconds_since_call_start;
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};
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struct vomp_sample_block {
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unsigned int codec;
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int len;
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time_ms_t starttime;
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time_ms_t endtime;
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unsigned char bytes[1024];
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};
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struct vomp_call_state {
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struct sched_ent alarm;
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struct vomp_call_half local;
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struct vomp_call_half remote;
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int initiated_call;
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int fast_audio;
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time_ms_t create_time;
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time_ms_t last_activity;
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time_ms_t audio_clock;
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int audio_started;
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// last local & remote status we sent to all interested parties
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int last_sent_status;
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unsigned char remote_codec_list[256];
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int recent_sample_rotor;
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struct vomp_sample_block recent_samples[VOMP_MAX_RECENT_SAMPLES];
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int sample_pos;
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unsigned int seen_samples[VOMP_MAX_RECENT_SAMPLES *4];
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};
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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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struct vomp_call_state vomp_call_states[VOMP_MAX_CALLS];
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struct profile_total vomp_stats;
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static void vomp_process_tick(struct sched_ent *alarm);
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static const char *vomp_describe_codec(int c);
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strbuf strbuf_append_vomp_supported_codecs(strbuf sb, const unsigned char supported_codecs[256]);
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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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struct 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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{
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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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if (debug & DEBUG_VOMP) DEBUGF("session=0x%08x",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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struct 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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{
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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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struct 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(struct 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=VOMP_STATE_NOCALL;
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call->remote.state=VOMP_STATE_NOCALL;
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call->last_sent_status=-1;
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call->create_time=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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if (debug & DEBUG_VOMP)
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DEBUGF("Returning new call #%d",local_session);
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return call;
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}
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struct 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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struct vomp_call_state *call;
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if (debug & DEBUG_VOMP)
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DEBUGF("%d calls already in progress.",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 (debug & DEBUG_VOMP)
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DEBUGF("asking for %06x:%06x, this call %06x:%06x",
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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 (debug & DEBUG_VOMP) {
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DEBUGF("%06x:%06x matches call #%d %06x:%06x",
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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);
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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(struct vomp_call_state *call)
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{
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overlay_mdp_frame mdp;
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unsigned short *len=&mdp.out.payload_length;
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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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time_ms_t call_millis = 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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*len=14;
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if (call->local.state < VOMP_STATE_RINGINGOUT && call->remote.state < VOMP_STATE_RINGINGOUT) {
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/* Include src and dst phone numbers */
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int didLen;
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/* Include the list of supported codecs */
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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[(*len)++]=i;
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if (debug & DEBUG_VOMP)
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DEBUGF("I support the %s codec", vomp_describe_codec(i));
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}
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mdp.out.payload[(*len)++]=0;
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if (call->initiated_call){
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DEBUGF("Sending phone numbers %s, %s",call->local.did,call->remote.did);
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didLen = snprintf((char *)(mdp.out.payload + *len), sizeof(mdp.out.payload) - *len, "%s", call->local.did);
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*len+=didLen+1;
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didLen = snprintf((char *)(mdp.out.payload + *len), sizeof(mdp.out.payload) - *len, "%s", call->remote.did);
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*len+=didLen+1;
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}
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if (debug & DEBUG_VOMP)
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DEBUGF("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) {
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unsigned char *p=&mdp.out.payload[0];
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struct 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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if (sb[rotor].len==vomp_sample_size(sb[rotor].codec)){
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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+sb[rotor].len) <VOMP_STUFF_BYTES && sb[rotor].len==vomp_sample_size(sb[rotor].codec)) {
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p[(*len)++]=sb[rotor].codec;
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bcopy(&sb[rotor].bytes[0],&p[*len],sb[rotor].len);
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(*len)+=sb[rotor].len;
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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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}
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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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// copy audio into the rotor buffers
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int vomp_received_audio(struct vomp_call_state *call, int audio_codec, const unsigned char *audio, int audio_length)
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{
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int codec_block_size=vomp_sample_size(audio_codec);
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int offset=0;
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struct vomp_sample_block *sb=call->recent_samples;
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while(offset<audio_length){
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int rotor=call->recent_sample_rotor%VOMP_MAX_RECENT_SAMPLES;
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if (sb[rotor].len==0 || call->audio_clock!=sb[rotor].endtime+1){
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/*
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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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sb[rotor].codec=audio_codec;
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sb[rotor].len=0;
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sb[rotor].starttime=call->audio_clock;
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sb[rotor].endtime=call->audio_clock+vomp_codec_timespan(sb[rotor].codec)-1;
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call->audio_clock=sb[rotor].endtime+1;
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}else if(sb[rotor].codec!=audio_codec){
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WHY("Did not finish previous audio buffer!!");
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}
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int copy_size = (audio_length - offset);
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if (copy_size > codec_block_size - sb[rotor].len)
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copy_size=codec_block_size - sb[rotor].len;
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bcopy(audio + offset,&sb[rotor].bytes[sb[rotor].len],copy_size);
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sb[rotor].len+=copy_size;
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offset+=copy_size;
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// send audio whenever we get the right number of bytes.
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if (sb[rotor].len>=codec_block_size){
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vomp_send_status_remote(call);
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}
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}
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return 0;
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}
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int monitor_call_status(struct vomp_call_state *call)
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{
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char msg[1024];
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int n = snprintf(msg,1024,"\nCALLSTATUS:%06x:%06x:%d:%d:%d:%s:%s:%s:%s\n",
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call->local.session,call->remote.session,
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call->local.state,call->remote.state,
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call->fast_audio,
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alloca_tohex_sid(call->local.sid),
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alloca_tohex_sid(call->remote.sid),
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call->local.did,call->remote.did);
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monitor_tell_clients(msg, n, MONITOR_VOMP);
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return 0;
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}
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int monitor_send_audio(struct vomp_call_state *call, int audio_codec, unsigned int start_time, unsigned int end_time, const unsigned char *audio, int audio_length, int sequence)
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{
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if (0) DEBUGF("Tell call monitor about audio for call %06x:%06x",
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call->local.session,call->remote.session);
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int sample_bytes=vomp_sample_size(audio_codec);
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char msg[1024 + MAX_AUDIO_BYTES];
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/* All commands followed by binary data start with *len:, so that
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they can be easily parsed at the far end, even if not supported.
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Put newline at start of these so that receiving data in command
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mode doesn't confuse the parser. */
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int msglen = snprintf(msg, 1024,
|
|
"\n*%d:AUDIOPACKET:%x:%d:%d:%d:%d\n",
|
|
sample_bytes,
|
|
call->local.session,
|
|
audio_codec, start_time, end_time,
|
|
sequence);
|
|
|
|
bcopy(audio, &msg[msglen], sample_bytes);
|
|
msglen+=sample_bytes;
|
|
msg[msglen++]='\n';
|
|
monitor_tell_clients(msg, msglen, MONITOR_VOMP);
|
|
return 0;
|
|
}
|
|
|
|
// update local state and notify interested clients with the correct message
|
|
int vomp_update_local_state(struct vomp_call_state *call, int new_state){
|
|
if (call->local.state>=new_state)
|
|
return 0;
|
|
|
|
switch(new_state){
|
|
case VOMP_STATE_CALLPREP:
|
|
// tell client our session id.
|
|
monitor_tell_formatted(MONITOR_VOMP, "\nCALLTO:%06x:%s:%s:%s:%s\n",
|
|
call->local.session,
|
|
alloca_tohex_sid(call->local.sid), call->local.did,
|
|
alloca_tohex_sid(call->remote.sid), call->remote.did);
|
|
break;
|
|
case VOMP_STATE_CALLENDED:
|
|
monitor_tell_formatted(MONITOR_VOMP, "\nHANGUP:%06x\n", call->local.session);
|
|
break;
|
|
}
|
|
|
|
call->local.state=new_state;
|
|
return 0;
|
|
}
|
|
|
|
// update remote state and notify interested clients with the correct message
|
|
int vomp_update_remote_state(struct vomp_call_state *call, int new_state){
|
|
if (call->remote.state>=new_state)
|
|
return 0;
|
|
|
|
switch(new_state){
|
|
case VOMP_STATE_RINGINGOUT:
|
|
monitor_tell_formatted(MONITOR_VOMP, "\nCALLFROM:%06x:%s:%s:%s:%s\n",
|
|
call->local.session,
|
|
alloca_tohex_sid(call->local.sid), call->local.did,
|
|
alloca_tohex_sid(call->remote.sid), call->remote.did);
|
|
break;
|
|
case VOMP_STATE_RINGINGIN:
|
|
monitor_tell_formatted(MONITOR_VOMP, "\nRINGING:%06x\n", call->local.session);
|
|
break;
|
|
case VOMP_STATE_INCALL:
|
|
if (call->remote.state==VOMP_STATE_RINGINGIN){
|
|
monitor_tell_formatted(MONITOR_VOMP, "\nANSWERED:%06x\n", call->local.session);
|
|
}
|
|
break;
|
|
}
|
|
|
|
call->remote.state=new_state;
|
|
return 0;
|
|
}
|
|
|
|
// send call state updates if required.
|
|
int vomp_update(struct vomp_call_state *call)
|
|
{
|
|
int combined_status=(call->remote.state<<4)|call->local.state;
|
|
|
|
if (call->last_sent_status==combined_status)
|
|
return 0;
|
|
|
|
if (debug & DEBUG_VOMP)
|
|
DEBUGF("Call state changed to %d %d, sending updates",call->local.state, call->remote.state);
|
|
|
|
call->last_sent_status=combined_status;
|
|
|
|
// tell the remote party
|
|
vomp_send_status_remote(call);
|
|
|
|
// tell monitor clients
|
|
if (monitor_socket_count && monitor_client_interested(MONITOR_VOMP))
|
|
monitor_call_status(call);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int vomp_call_start_audio(struct vomp_call_state *call)
|
|
{
|
|
call->audio_started=1;
|
|
return WHY("Not implemented");
|
|
}
|
|
|
|
// check a small circular buffer of recently seen audio
|
|
// we're not trying to be perfect here, we still expect all clients to reorder and filter duplicates
|
|
int vomp_audio_already_seen(struct vomp_call_state *call, unsigned int end_time)
|
|
{
|
|
int i;
|
|
for(i=0;i<VOMP_MAX_RECENT_SAMPLES *4;i++)
|
|
if (call->seen_samples[i]==end_time)
|
|
return 1;
|
|
call->seen_samples[call->sample_pos]=end_time;
|
|
call->sample_pos++;
|
|
if (call->sample_pos>=VOMP_MAX_RECENT_SAMPLES *4)
|
|
call->sample_pos=0;
|
|
return 0;
|
|
}
|
|
|
|
int vomp_process_audio(struct vomp_call_state *call,unsigned int sender_duration,overlay_mdp_frame *mdp)
|
|
{
|
|
int ofs=14;
|
|
// if (mdp->in.payload_length>14)
|
|
// DEBUGF("got here (payload has %d bytes)",mdp->in.payload_length);
|
|
|
|
/* Get end time marker for sample block collection */
|
|
unsigned int e=0, s=0;
|
|
e=mdp->in.payload[ofs++]<<24;
|
|
e|=mdp->in.payload[ofs++]<<16;
|
|
e|=mdp->in.payload[ofs++]<<8;
|
|
e|=mdp->in.payload[ofs++]<<0;
|
|
|
|
sender_duration = (e&0xFFFF0000)|sender_duration;
|
|
if (debug & DEBUG_VOMP)
|
|
DEBUGF("Jitter %d, %lld", sender_duration - e, (long long)((gettime_ms() - call->create_time) - e));
|
|
|
|
int sequence = call->remote.sequence;
|
|
|
|
while(ofs<mdp->in.payload_length)
|
|
{
|
|
int codec=mdp->in.payload[ofs];
|
|
// DEBUGF("Spotted a %s sample block",vomp_describe_codec(codec));
|
|
if (!codec||vomp_sample_size(codec)<0) break;
|
|
if ((ofs+1+vomp_sample_size(codec))>mdp->in.payload_length) break;
|
|
|
|
/* work out start-time from end-time less duration of included sample(s).
|
|
XXX - Assumes only non-adaptive codecs. */
|
|
s = e-vomp_codec_timespan(codec)+1;
|
|
|
|
/* Pass audio frame to all registered listeners */
|
|
if (!vomp_audio_already_seen(call, e)){
|
|
if (monitor_socket_count)
|
|
monitor_send_audio(call, codec, s, e,
|
|
&mdp->in.payload[ofs+1],
|
|
vomp_sample_size(codec),
|
|
sequence);
|
|
}
|
|
ofs+=1+vomp_sample_size(codec);
|
|
e=s-1;
|
|
sequence--;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int vomp_call_stop_audio(struct vomp_call_state *call)
|
|
{
|
|
call->audio_started=0;
|
|
return WHY("Not implemented");
|
|
}
|
|
|
|
int vomp_ringing(struct vomp_call_state *call){
|
|
if (call){
|
|
if ((!call->initiated_call) && call->local.state<VOMP_STATE_RINGINGIN && call->remote.state==VOMP_STATE_RINGINGOUT){
|
|
if (debug & DEBUG_VOMP)
|
|
DEBUGF("RING RING!");
|
|
vomp_update_local_state(call, VOMP_STATE_RINGINGIN);
|
|
vomp_update(call);
|
|
}else
|
|
return WHY("Can't ring, call is not being dialled");
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int vomp_call_destroy(struct vomp_call_state *call)
|
|
{
|
|
/* do some general clean ups */
|
|
if (call->audio_started) vomp_call_stop_audio(call);
|
|
|
|
if (debug & DEBUG_VOMP)
|
|
DEBUGF("Destroying call %s <--> %s", call->local.did,call->remote.did);
|
|
|
|
/* tell everyone the call has died */
|
|
vomp_update_local_state(call, 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(struct 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.
|
|
*/
|
|
if (debug & DEBUG_VOMP)
|
|
DEBUG("Dialing");
|
|
|
|
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 */
|
|
struct vomp_call_state *call=vomp_create_call(
|
|
remote_sid,
|
|
local_sid,
|
|
0,
|
|
0);
|
|
|
|
/* Copy local / remote phone numbers */
|
|
strlcpy(call->local.did, local_did, sizeof(call->local.did));
|
|
strlcpy(call->remote.did, remote_did, sizeof(call->remote.did));
|
|
|
|
vomp_update_local_state(call, 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(struct vomp_call_state *call)
|
|
{
|
|
if (call){
|
|
if (debug & DEBUG_VOMP)
|
|
DEBUG("Picking up");
|
|
if (call->local.state<=VOMP_STATE_RINGINGIN && call->remote.state==VOMP_STATE_RINGINGOUT){
|
|
vomp_update_local_state(call, VOMP_STATE_INCALL);
|
|
call->create_time=gettime_ms();
|
|
/* state machine does job of starting audio stream, just tell everyone about
|
|
the changed state. */
|
|
vomp_update(call);
|
|
}else
|
|
return WHY("Can't pickup, call is not ringing");
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int vomp_hangup(struct vomp_call_state *call)
|
|
{
|
|
if (call){
|
|
if (debug & DEBUG_VOMP)
|
|
DEBUG("Hanging up");
|
|
if (call->local.state==VOMP_STATE_INCALL) vomp_call_stop_audio(call);
|
|
vomp_update_local_state(call, VOMP_STATE_CALLENDED);
|
|
vomp_update(call);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int vomp_extract_remote_codec_list(struct vomp_call_state *call,overlay_mdp_frame *mdp)
|
|
{
|
|
int ofs=14;
|
|
|
|
if (debug & DEBUG_VOMP)
|
|
dump("codec list mdp frame", (unsigned char *)&mdp->in.payload[0],mdp->in.payload_length);
|
|
|
|
for (;ofs<mdp->in.payload_length && mdp->in.payload[ofs];ofs++){
|
|
call->remote_codec_list[mdp->in.payload[ofs]]=1;
|
|
}
|
|
if (!call->initiated_call){
|
|
ofs++;
|
|
if (ofs<mdp->in.payload_length)
|
|
ofs+=strlcpy(call->remote.did, (char *)(mdp->in.payload+ofs), sizeof(call->remote.did))+1;
|
|
if (ofs<mdp->in.payload_length)
|
|
ofs+=strlcpy(call->local.did, (char *)(mdp->in.payload+ofs), sizeof(call->local.did));
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/* 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 */
|
|
|
|
struct 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 && (debug & DEBUG_VOMP))
|
|
DEBUG("recvr_session==0, created call");
|
|
|
|
recvr_state = call->local.state;
|
|
call->remote.sequence=sender_seq;
|
|
|
|
|
|
// TODO ignore state changes if sequence is stale?
|
|
// TODO ignore state changes that seem to go backwards?
|
|
|
|
if ((!monitor_socket_count)
|
|
&&(!monitor_client_interested(MONITOR_VOMP)))
|
|
{
|
|
/* No registered listener, so we cannot answer the call, so just reject
|
|
it. */
|
|
if (debug & DEBUG_VOMP)
|
|
DEBUGF("Rejecting call due to lack of a listener: states=%d,%d", recvr_state, sender_state);
|
|
|
|
recvr_state=VOMP_STATE_CALLENDED;
|
|
/* now let the state machine progress to destroy the call */
|
|
}
|
|
|
|
if (recvr_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);
|
|
recvr_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=recvr_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.
|
|
recvr_state=VOMP_STATE_CALLENDED;
|
|
else{
|
|
// Don't automatically transition to RINGIN, wait for a client to tell us when.
|
|
}
|
|
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
|
|
recvr_state=VOMP_STATE_RINGINGOUT;
|
|
}else{
|
|
recvr_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 */
|
|
recvr_state=VOMP_STATE_INCALL;
|
|
// reset create time when call is established
|
|
call->create_time=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 */
|
|
recvr_state=VOMP_STATE_INCALL;
|
|
// reset create time when call is established
|
|
call->create_time=gettime_ms();
|
|
// 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.
|
|
*/
|
|
WHYF("Ignoring invalid call state %d.%d",sender_state,recvr_state);
|
|
return 0;
|
|
}
|
|
|
|
vomp_update_remote_state(call, sender_state);
|
|
vomp_update_local_state(call, recvr_state);
|
|
call->last_activity=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 (sender_state==VOMP_STATE_CALLENDED
|
|
&&recvr_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?");
|
|
}
|
|
|
|
static const 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 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 '?';
|
|
}
|
|
|
|
static void vomp_process_tick(struct sched_ent *alarm)
|
|
{
|
|
char msg[32];
|
|
int len;
|
|
time_ms_t now = gettime_ms();
|
|
|
|
struct vomp_call_state *call = (struct vomp_call_state *)alarm;
|
|
|
|
/* See if any calls need to be expired.
|
|
Allow VOMP_CALL_DIAL_TIMEOUT ms for the other party to ring / request ringing
|
|
Allow VOMP_CALL_RING_TIMEOUT ms for the ringing party to answer
|
|
Allow VOMP_CALL_NETWORK_TIMEOUT ms between received packets
|
|
*/
|
|
|
|
if ((call->remote.state < VOMP_STATE_RINGINGOUT && call->create_time + VOMP_CALL_DIAL_TIMEOUT < now) ||
|
|
(call->local.state < VOMP_STATE_INCALL && call->create_time + VOMP_CALL_RING_TIMEOUT < now) ||
|
|
(call->last_activity+VOMP_CALL_NETWORK_TIMEOUT<now) ){
|
|
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 = gettime_ms() + VOMP_CALL_STATUS_INTERVAL;
|
|
alarm->deadline = alarm->alarm + VOMP_CALL_STATUS_INTERVAL/2;
|
|
schedule(alarm);
|
|
}
|