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serval-dna/vomp.c
Jeremy Lakeman 66e0711d6b Only send one audio block per packet, but request that it be sent multiple times 
If we receive a large buffer of audio, we want to stuff the packet with multiple frames and send them together.
And we want to send redundant copies of the audio to help recover from packet loss.
But if all our redundant copies end up in the same packet, we're screwed anyway.

This is a temporary hack until the network layer implements NACK / retry for resilient multi-hop delivery
2012-09-28 15:23:49 +09:30

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/*
Serval Voice Over Mesh Protocol (VoMP)
Copyright (C) 2012 Paul Gardner-Stephen
Copyright (C) 2012 Serval Project Inc.
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*/
/*
VoMP works using a 6-state model of a phone call, and relies on MDP for
auth-cryption of frames. VoMP provides it's own replay protection.
*/
#include "serval.h"
#include "strbuf.h"
#include "strlcpy.h"
/*
Typical call state lifecycle between 2 parties.
Legend;
# incoming command from monitor client
$ outgoing monitor status
<> vomp packet with state change sent across the network
Monitor Init
# MONITOR VOMP [supported codec list]
Dialing
// client requests an outgoing call
# CALL [sid] [myDid] [TheirDid]
> CALLPREP + codecs + phone numbers
// let the client know what token we are going to use for the remainder of the call
$ CALLTO [token] [mySid] [myDid] [TheirSid] [TheirDid]
// allocate a session number and tell them our codecs,
// but we don't need to do anything else yet,
// this might be a replay attack
< NOCALL + codecs
// Ok, we have a network path, lets try to establish the call
> RINGOUT
// (Note that if both parties are trying to dial each other,
// the call should jump straight to INCALL)
// inform client about the call request
$ CALLFROM [token] [mySid] [myDid] [TheirSid] [TheirDid]
// Note that we may need to wait for other external processes
// before a phone is actually ringing
# RING [token]
< RINGIN
// All good, there's a phone out there ringing, you can indicate that to the user
$ RINGING [token]
Answering
# PICKUP [token]
< INCALL
// The client can now start sending audio
> INCALL
$ INCALL [token]
// The client can now start sending audio
$ INCALL [token]
Tell any clients that the call hasn't timed out yet
(if servald is behaving this should be redundant, if it isn't behaving how do we hangup?)
$ KEEPALIVE [token]
Hanging up (may also be triggered on network or call establishment timeout)
# HANGUP [token]
> CALLENDED
$ HANGUP [token]
< CALLENDED
$ HANGUP [token]
*/
/*
Minimum network format requirements;
- your call session, packed integer
- my call state
- my sequence number
Pre-ring call setup;
- my call session
- my supported codec list
- your number
- my number
- my name
In call audio;
- codec
- elapsed time from call start
- audio duration
- audio data (remainder of payload)
Assuming minimum audio duration per packet is 20ms, 1 byte sequence should let us deal with ~2.5s of jitter.
If we have >2.5s of jitter, the network is obviously too crappy to support a voice call anyway.
If we can assume constant duration per codec, and I believe we can,
we can use the sequence number to derive the other audio timing information.
We need to resume a call even with large periods of zero traffic (eg >10s),
we should be able to use our own wall clock to estimate which 5s interval the audio belongs to.
*/
// ideally these id's should only be used on the network, with monitor events to inform clients of state changes
#define VOMP_STATE_NOCALL 1
#define VOMP_STATE_CALLPREP 2
#define VOMP_STATE_RINGINGOUT 3
#define VOMP_STATE_RINGINGIN 4
#define VOMP_STATE_INCALL 5
#define VOMP_STATE_CALLENDED 6
#define VOMP_SESSION_MASK 0xffffff
#define VOMP_MAX_CALLS 16
struct vomp_call_half {
unsigned char sid[SID_SIZE];
char did[64];
unsigned char state;
unsigned char codec;
unsigned int session;
unsigned int sequence;
};
struct vomp_call_state {
struct sched_ent alarm;
struct vomp_call_half local;
struct vomp_call_half remote;
int initiated_call;
time_ms_t create_time;
time_ms_t last_activity;
time_ms_t audio_clock;
// last local & remote status we sent to all interested parties
int last_sent_status;
unsigned char remote_codec_list[256];
// track when we first heard audio, so we can calculate timing from the current sequence number
int first_remote_audio_sequence;
// simple ring buffer of audio sample times, used to drop duplicate incoming frames
// stores end times, since this is an odd number we can initialise the buffer to zero's
int sample_pos;
unsigned int seen_samples[VOMP_MAX_RECENT_SAMPLES *4];
};
/* Some clients may only support one call at a time, even then we allow for multiple call states.
This is partly to deal with denial of service attacks that might occur by causing
the ejection of newly allocated session numbers before the caller has had a chance
to progress the call to a further state. */
int vomp_call_count=0;
struct vomp_call_state vomp_call_states[VOMP_MAX_CALLS];
struct profile_total vomp_stats;
static void vomp_process_tick(struct sched_ent *alarm);
static const char *vomp_describe_codec(int c);
strbuf strbuf_append_vomp_supported_codecs(strbuf sb, const unsigned char supported_codecs[256]);
/* which codecs we support (set by registered listener) */
unsigned char vomp_local_codec_list[256];
struct vomp_call_state *vomp_find_call_by_session(int session_token)
{
int i;
for(i=0;i<vomp_call_count;i++)
if (session_token==vomp_call_states[i].local.session)
return &vomp_call_states[i];
return NULL;
}
int vomp_generate_session_id()
{
int session_id=0;
while (!session_id)
{
if (urandombytes((unsigned char *)&session_id,sizeof(int)))
return WHY("Insufficient entropy");
session_id&=VOMP_SESSION_MASK;
if (debug & DEBUG_VOMP) DEBUGF("session=0x%08x",session_id);
int i;
/* reject duplicate call session numbers */
for(i=0;i<vomp_call_count;i++)
if (session_id==vomp_call_states[i].local.session
||session_id==vomp_call_states[i].local.session){
session_id=0;
break;
}
}
return session_id;
}
struct vomp_call_state *vomp_create_call(unsigned char *remote_sid,
unsigned char *local_sid,
unsigned int remote_session,
unsigned int local_session)
{
if (!local_session)
local_session=vomp_generate_session_id();
struct vomp_call_state *call = &vomp_call_states[vomp_call_count];
vomp_call_count++;
/* prepare slot */
bzero(call,sizeof(struct vomp_call_state));
bcopy(local_sid,call->local.sid,SID_SIZE);
bcopy(remote_sid,call->remote.sid,SID_SIZE);
call->local.session=local_session;
call->remote.session=remote_session;
call->local.state=VOMP_STATE_NOCALL;
call->remote.state=VOMP_STATE_NOCALL;
call->last_sent_status=-1;
call->create_time=gettime_ms();
call->last_activity=call->create_time;
call->alarm.alarm = call->create_time+VOMP_CALL_STATUS_INTERVAL;
call->alarm.function = vomp_process_tick;
vomp_stats.name="vomp_process_tick";
call->alarm.stats=&vomp_stats;
schedule(&call->alarm);
if (debug & DEBUG_VOMP)
DEBUGF("Returning new call #%d",local_session);
return call;
}
struct vomp_call_state *vomp_find_or_create_call(unsigned char *remote_sid,
unsigned char *local_sid,
unsigned int sender_session,
unsigned int recvr_session,
int sender_state,
int recvr_state)
{
int i;
struct vomp_call_state *call;
if (debug & DEBUG_VOMP)
DEBUGF("%d calls already in progress.",vomp_call_count);
for(i=0;i<vomp_call_count;i++)
{
call = &vomp_call_states[i];
/* do the fast comparison first, and only if that matches proceed to
the slower SID comparisons */
if (debug & DEBUG_VOMP)
DEBUGF("asking for %06x:%06x, this call %06x:%06x",
sender_session,recvr_session,
call->remote.session,
call->local.session);
int checked=0;
if (call->remote.session&&sender_session) {
checked++;
if(sender_session!=call->remote.session)
continue;
}
if (call->local.session&&recvr_session) {
checked++;
if(recvr_session!=call->local.session)
continue;
}
if (!checked) continue;
if (memcmp(remote_sid,call->remote.sid,SID_SIZE)) continue;
if (memcmp(local_sid,call->local.sid,SID_SIZE)) continue;
/* it matches. */
/* Record session number if required */
if (!call->remote.session)
call->remote.session=sender_session;
if (debug & DEBUG_VOMP) {
DEBUGF("%06x:%06x matches call #%d %06x:%06x",
sender_session,recvr_session,i,
call->remote.session,
call->local.session);
}
return call;
}
/* Don't create a call record if either party has ended it */
if (sender_state==VOMP_STATE_CALLENDED || recvr_state==VOMP_STATE_CALLENDED)
return NULL;
/* Only create a call record if either party is in CALLPREP state */
if (sender_state==VOMP_STATE_CALLPREP || recvr_state==VOMP_STATE_CALLPREP)
return vomp_create_call(remote_sid, local_sid, sender_session, recvr_session);
return NULL;
}
static void prepare_vomp_header(struct vomp_call_state *call, overlay_mdp_frame *mdp){
mdp->packetTypeAndFlags=MDP_TX;
bcopy(call->local.sid,mdp->out.src.sid,SID_SIZE);
mdp->out.src.port=MDP_PORT_VOMP;
bcopy(call->remote.sid,mdp->out.dst.sid,SID_SIZE);
mdp->out.dst.port=MDP_PORT_VOMP;
mdp->out.payload[0]=0x01; /* Normal VoMP frame */
mdp->out.payload[1]=(call->remote.state<<4)|call->local.state;
mdp->out.payload[2]=(call->remote.sequence>>8)&0xff;
mdp->out.payload[3]=(call->remote.sequence>>0)&0xff;
mdp->out.payload[4]=(call->local.sequence>>8)&0xff;
mdp->out.payload[5]=(call->local.sequence>>0)&0xff;
time_ms_t call_millis = gettime_ms() - call->create_time;
mdp->out.payload[6]=(call_millis>>8)&0xff;
mdp->out.payload[7]=(call_millis>>0)&0xff;
mdp->out.payload[8]=(call->remote.session>>16)&0xff;
mdp->out.payload[9]=(call->remote.session>>8)&0xff;
mdp->out.payload[10]=(call->remote.session>>0)&0xff;
mdp->out.payload[11]=(call->local.session>>16)&0xff;
mdp->out.payload[12]=(call->local.session>>8)&0xff;
mdp->out.payload[13]=(call->local.session>>0)&0xff;
mdp->out.payload_length=14;
}
/* send updated call status to end-point and to any interested listeners as
appropriate */
int vomp_send_status_remote(struct vomp_call_state *call)
{
overlay_mdp_frame mdp;
unsigned short *len=&mdp.out.payload_length;
bzero(&mdp,sizeof(mdp));
prepare_vomp_header(call, &mdp);
if (call->local.state < VOMP_STATE_RINGINGOUT && call->remote.state < VOMP_STATE_RINGINGOUT) {
/* Include src and dst phone numbers */
int didLen;
/* Include the list of supported codecs */
int i;
for (i = 0; i < 256; ++i)
if (vomp_local_codec_list[i]) {
mdp.out.payload[(*len)++]=i;
if (debug & DEBUG_VOMP)
DEBUGF("I support the %s codec", vomp_describe_codec(i));
}
mdp.out.payload[(*len)++]=0;
if (call->initiated_call){
DEBUGF("Sending phone numbers %s, %s",call->local.did,call->remote.did);
didLen = snprintf((char *)(mdp.out.payload + *len), sizeof(mdp.out.payload) - *len, "%s", call->local.did);
*len+=didLen+1;
didLen = snprintf((char *)(mdp.out.payload + *len), sizeof(mdp.out.payload) - *len, "%s", call->remote.did);
*len+=didLen+1;
}
if (debug & DEBUG_VOMP)
DEBUGF("mdp frame with codec list is %d bytes", mdp.out.payload_length);
}
overlay_mdp_dispatch(&mdp,0,NULL,0);
call->local.sequence++;
return 0;
}
// copy audio into the rotor buffers
int vomp_received_audio(struct vomp_call_state *call, int audio_codec, const unsigned char *audio, int audio_length)
{
if (call->local.state!=VOMP_STATE_INCALL)
return -1;
int codec_block_size=vomp_sample_size(audio_codec);
int offset=0;
int codec_duration = vomp_codec_timespan(audio_codec);
while(offset<audio_length){
overlay_mdp_frame mdp;
unsigned short *len=&mdp.out.payload_length;
bzero(&mdp,sizeof(mdp));
prepare_vomp_header(call, &mdp);
/*
Note that in-call slew is the responsibility of the player, not the
recorder of audio. Basically if the audio queue starts to bank up,
then the player needs to drop samples.
*/
mdp.out.payload[(*len)++]=(call->audio_clock>>24)&0xff;
mdp.out.payload[(*len)++]=(call->audio_clock>>16)&0xff;
mdp.out.payload[(*len)++]=(call->audio_clock>>8)&0xff;
mdp.out.payload[(*len)++]=(call->audio_clock>>0)&0xff;
mdp.out.payload[(*len)++]=audio_codec;
if (offset+codec_block_size>audio_length)
codec_block_size = audio_length - offset;
bcopy(audio+offset,&mdp.out.payload[(*len)],codec_block_size);
(*len)+=codec_block_size;
offset+=codec_block_size;
call->audio_clock += codec_duration;
// send the payload more than once to add resilience to dropped packets
// TODO remove once network links have built in retries
mdp.out.send_copies=VOMP_MAX_RECENT_SAMPLES;
overlay_mdp_dispatch(&mdp,0,NULL,0);
call->local.sequence++;
}
return 0;
}
int monitor_call_status(struct vomp_call_state *call)
{
char msg[1024];
int n = snprintf(msg,1024,"\nCALLSTATUS:%06x:%06x:%d:%d:%d:%s:%s:%s:%s\n",
call->local.session,call->remote.session,
call->local.state,call->remote.state,
0,
alloca_tohex_sid(call->local.sid),
alloca_tohex_sid(call->remote.sid),
call->local.did,call->remote.did);
monitor_tell_clients(msg, n, MONITOR_VOMP);
return 0;
}
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)
{
if (0) DEBUGF("Tell call monitor about audio for call %06x:%06x",
call->local.session,call->remote.session);
int sample_bytes=vomp_sample_size(audio_codec);
char msg[1024 + MAX_AUDIO_BYTES];
/* All commands followed by binary data start with *len:, so that
they can be easily parsed at the far end, even if not supported.
Put newline at start of these so that receiving data in command
mode doesn't confuse the parser. */
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;
}
// 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;
int sequence = call->remote.sequence;
if(ofs<mdp->in.payload_length)
{
s=mdp->in.payload[ofs++]<<24;
s|=mdp->in.payload[ofs++]<<16;
s|=mdp->in.payload[ofs++]<<8;
s|=mdp->in.payload[ofs++]<<0;
sender_duration = (s&0xFFFF0000)|sender_duration;
// simplistic jitter debug info
if (debug & DEBUG_VOMP)
DEBUGF("Jitter %d, %lld", sender_duration - s, (long long)((gettime_ms() - call->create_time) - s));
int codec=mdp->in.payload[ofs++];
int audio_len = mdp->in.payload_length - ofs;
if ((!codec)||vomp_sample_size(codec)<0) return -1;
e = s + 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],
audio_len,
sequence);
}
}
return 0;
}
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)
{
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, const char *local_did, const 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");
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 */
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. */
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);
}
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