mirror of
https://github.com/danog/libtgvoip.git
synced 2024-11-30 04:39:03 +01:00
2537 lines
77 KiB
C++
2537 lines
77 KiB
C++
//
|
|
// libtgvoip is free and unencumbered public domain software.
|
|
// For more information, see http://unlicense.org or the UNLICENSE file
|
|
// you should have received with this source code distribution.
|
|
//
|
|
|
|
#ifndef _WIN32
|
|
#include <unistd.h>
|
|
#include <sys/time.h>
|
|
#endif
|
|
#include <errno.h>
|
|
#include <string.h>
|
|
#include <wchar.h>
|
|
#include "VoIPController.h"
|
|
#include "logging.h"
|
|
#include "threading.h"
|
|
#include "BufferOutputStream.h"
|
|
#include "BufferInputStream.h"
|
|
#include "OpusEncoder.h"
|
|
#include "OpusDecoder.h"
|
|
#include "VoIPServerConfig.h"
|
|
#include <assert.h>
|
|
#include <time.h>
|
|
#include <math.h>
|
|
#include <exception>
|
|
#include <stdexcept>
|
|
#include <algorithm>
|
|
|
|
|
|
#define PKT_INIT 1
|
|
#define PKT_INIT_ACK 2
|
|
#define PKT_STREAM_STATE 3
|
|
#define PKT_STREAM_DATA 4
|
|
#define PKT_UPDATE_STREAMS 5
|
|
#define PKT_PING 6
|
|
#define PKT_PONG 7
|
|
#define PKT_STREAM_DATA_X2 8
|
|
#define PKT_STREAM_DATA_X3 9
|
|
#define PKT_LAN_ENDPOINT 10
|
|
#define PKT_NETWORK_CHANGED 11
|
|
#define PKT_SWITCH_PREF_RELAY 12
|
|
#define PKT_SWITCH_TO_P2P 13
|
|
#define PKT_NOP 14
|
|
|
|
#define IS_MOBILE_NETWORK(x) (x==NET_TYPE_GPRS || x==NET_TYPE_EDGE || x==NET_TYPE_3G || x==NET_TYPE_HSPA || x==NET_TYPE_LTE || x==NET_TYPE_OTHER_MOBILE)
|
|
|
|
#define PROTOCOL_NAME 0x50567247 // "GrVP" in little endian (reversed here)
|
|
#define PROTOCOL_VERSION 3
|
|
#define MIN_PROTOCOL_VERSION 3
|
|
|
|
#define STREAM_DATA_FLAG_LEN16 0x40
|
|
#define STREAM_DATA_FLAG_HAS_MORE_FLAGS 0x80
|
|
|
|
#define STREAM_TYPE_AUDIO 1
|
|
#define STREAM_TYPE_VIDEO 2
|
|
|
|
#define CODEC_OPUS 1
|
|
|
|
/*flags:# voice_call_id:flags.2?int128 in_seq_no:flags.4?int out_seq_no:flags.4?int
|
|
* recent_received_mask:flags.5?int proto:flags.3?int extra:flags.1?string raw_data:flags.0?string*/
|
|
#define PFLAG_HAS_DATA 1
|
|
#define PFLAG_HAS_EXTRA 2
|
|
#define PFLAG_HAS_CALL_ID 4
|
|
#define PFLAG_HAS_PROTO 8
|
|
#define PFLAG_HAS_SEQ 16
|
|
#define PFLAG_HAS_RECENT_RECV 32
|
|
|
|
#define INIT_FLAG_DATA_SAVING_ENABLED 1
|
|
|
|
#define TLID_DECRYPTED_AUDIO_BLOCK 0xDBF948C1
|
|
#define TLID_SIMPLE_AUDIO_BLOCK 0xCC0D0E76
|
|
#define TLID_UDP_REFLECTOR_PEER_INFO 0x27D9371C
|
|
#define TLID_UDP_REFLECTOR_PEER_INFO_IPV6 0x83fc73b1
|
|
#define TLID_UDP_REFLECTOR_SELF_INFO 0xc01572c7
|
|
#define PAD4(x) (4-(x+(x<=253 ? 1 : 0))%4)
|
|
|
|
#define MAX(a,b) (a>b ? a : b)
|
|
#define MIN(a,b) (a<b ? a : b)
|
|
|
|
inline int pad4(int x){
|
|
int r=PAD4(x);
|
|
if(r==4)
|
|
return 0;
|
|
return r;
|
|
}
|
|
|
|
|
|
using namespace tgvoip;
|
|
|
|
#ifdef __APPLE__
|
|
#include "os/darwin/AudioUnitIO.h"
|
|
#include <mach/mach_time.h>
|
|
double VoIPController::machTimebase=0;
|
|
uint64_t VoIPController::machTimestart=0;
|
|
#endif
|
|
|
|
#ifdef _WIN32
|
|
int64_t VoIPController::win32TimeScale = 0;
|
|
bool VoIPController::didInitWin32TimeScale = false;
|
|
#endif
|
|
|
|
#define SHA1_LENGTH 20
|
|
#define SHA256_LENGTH 32
|
|
|
|
#ifndef TGVOIP_USE_CUSTOM_CRYPTO
|
|
#include <openssl/sha.h>
|
|
#include <openssl/aes.h>
|
|
#include <openssl/modes.h>
|
|
#include <openssl/rand.h>
|
|
|
|
void tgvoip_openssl_aes_ige_encrypt(uint8_t* in, uint8_t* out, size_t length, uint8_t* key, uint8_t* iv){
|
|
AES_KEY akey;
|
|
AES_set_encrypt_key(key, 32*8, &akey);
|
|
AES_ige_encrypt(in, out, length, &akey, iv, AES_ENCRYPT);
|
|
}
|
|
|
|
void tgvoip_openssl_aes_ige_decrypt(uint8_t* in, uint8_t* out, size_t length, uint8_t* key, uint8_t* iv){
|
|
AES_KEY akey;
|
|
AES_set_decrypt_key(key, 32*8, &akey);
|
|
AES_ige_encrypt(in, out, length, &akey, iv, AES_DECRYPT);
|
|
}
|
|
|
|
void tgvoip_openssl_rand_bytes(uint8_t* buffer, size_t len){
|
|
RAND_bytes(buffer, len);
|
|
}
|
|
|
|
void tgvoip_openssl_sha1(uint8_t* msg, size_t len, uint8_t* output){
|
|
SHA1(msg, len, output);
|
|
}
|
|
|
|
void tgvoip_openssl_sha256(uint8_t* msg, size_t len, uint8_t* output){
|
|
SHA256(msg, len, output);
|
|
}
|
|
|
|
void tgvoip_openssl_aes_ctr_encrypt(uint8_t* inout, size_t length, uint8_t* key, uint8_t* iv, uint8_t* ecount, uint32_t* num){
|
|
AES_KEY akey;
|
|
AES_set_encrypt_key(key, 32*8, &akey);
|
|
CRYPTO_ctr128_encrypt(inout, inout, length, &akey, iv, ecount, num, (block128_f) AES_encrypt);
|
|
}
|
|
|
|
voip_crypto_functions_t VoIPController::crypto={
|
|
tgvoip_openssl_rand_bytes,
|
|
tgvoip_openssl_sha1,
|
|
tgvoip_openssl_sha256,
|
|
tgvoip_openssl_aes_ige_encrypt,
|
|
tgvoip_openssl_aes_ige_decrypt,
|
|
tgvoip_openssl_aes_ctr_encrypt
|
|
|
|
};
|
|
#else
|
|
voip_crypto_functions_t VoIPController::crypto; // set it yourself upon initialization
|
|
#endif
|
|
|
|
#ifdef _MSC_VER
|
|
#define MSC_STACK_FALLBACK(a, b) (b)
|
|
#else
|
|
#define MSC_STACK_FALLBACK(a, b) (a)
|
|
#endif
|
|
|
|
extern FILE* tgvoipLogFile;
|
|
|
|
VoIPController::VoIPController() : activeNetItfName(""),
|
|
currentAudioInput("default"),
|
|
currentAudioOutput("default"),
|
|
proxyAddress(""),
|
|
proxyUsername(""),
|
|
proxyPassword(""),
|
|
outgoingPacketsBufferPool(1024, 1024){
|
|
seq=1;
|
|
lastRemoteSeq=0;
|
|
state=STATE_WAIT_INIT;
|
|
audioInput=NULL;
|
|
audioOutput=NULL;
|
|
decoder=NULL;
|
|
encoder=NULL;
|
|
jitterBuffer=NULL;
|
|
audioOutStarted=false;
|
|
audioTimestampIn=0;
|
|
audioTimestampOut=0;
|
|
stopping=false;
|
|
int i;
|
|
sendQueue=new BlockingQueue<PendingOutgoingPacket>(21);
|
|
init_mutex(sendBufferMutex);
|
|
memset(remoteAcks, 0, sizeof(double)*32);
|
|
memset(sentPacketTimes, 0, sizeof(double)*32);
|
|
memset(recvPacketTimes, 0, sizeof(double)*32);
|
|
memset(rttHistory, 0, sizeof(double)*32);
|
|
memset(sendLossCountHistory, 0, sizeof(uint32_t)*32);
|
|
memset(&stats, 0, sizeof(voip_stats_t));
|
|
lastRemoteAckSeq=0;
|
|
lastSentSeq=0;
|
|
recvLossCount=0;
|
|
packetsRecieved=0;
|
|
waitingForAcks=false;
|
|
networkType=NET_TYPE_UNKNOWN;
|
|
stateCallback=NULL;
|
|
echoCanceller=NULL;
|
|
dontSendPackets=0;
|
|
micMuted=false;
|
|
currentEndpoint=NULL;
|
|
waitingForRelayPeerInfo=false;
|
|
allowP2p=true;
|
|
dataSavingMode=false;
|
|
publicEndpointsReqTime=0;
|
|
init_mutex(queuedPacketsMutex);
|
|
init_mutex(endpointsMutex);
|
|
connectionInitTime=0;
|
|
lastRecvPacketTime=0;
|
|
dataSavingRequestedByPeer=false;
|
|
peerVersion=0;
|
|
conctl=new CongestionControl();
|
|
prevSendLossCount=0;
|
|
receivedInit=false;
|
|
receivedInitAck=false;
|
|
peerPreferredRelay=NULL;
|
|
preferredRelay=NULL;
|
|
statsDump=NULL;
|
|
useTCP=false;
|
|
useUDP=true;
|
|
didAddTcpRelays=false;
|
|
setEstablishedAt=0;
|
|
udpPingCount=0;
|
|
lastUdpPingTime=0;
|
|
openingTcpSocket=NULL;
|
|
|
|
proxyProtocol=PROXY_NONE;
|
|
proxyPort=0;
|
|
resolvedProxyAddress=NULL;
|
|
|
|
signalBarCount=0;
|
|
signalBarCountCallback=NULL;
|
|
|
|
selectCanceller=SocketSelectCanceller::Create();
|
|
udpSocket=NetworkSocket::Create(PROTO_UDP);
|
|
realUdpSocket=udpSocket;
|
|
udpConnectivityState=UDP_UNKNOWN;
|
|
echoCancellationStrength=1;
|
|
|
|
outputAGC=NULL;
|
|
outputAGCEnabled=false;
|
|
|
|
maxAudioBitrate=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_max_bitrate", 20000);
|
|
maxAudioBitrateGPRS=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_max_bitrate_gprs", 8000);
|
|
maxAudioBitrateEDGE=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_max_bitrate_edge", 16000);
|
|
maxAudioBitrateSaving=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_max_bitrate_saving", 8000);
|
|
initAudioBitrate=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_init_bitrate", 16000);
|
|
initAudioBitrateGPRS=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_init_bitrate_gprs", 8000);
|
|
initAudioBitrateEDGE=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_init_bitrate_edge", 8000);
|
|
initAudioBitrateSaving=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_init_bitrate_saving", 8000);
|
|
audioBitrateStepIncr=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_bitrate_step_incr", 1000);
|
|
audioBitrateStepDecr=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_bitrate_step_decr", 1000);
|
|
minAudioBitrate=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_min_bitrate", 8000);
|
|
relaySwitchThreshold=ServerConfig::GetSharedInstance()->GetDouble("relay_switch_threshold", 0.8);
|
|
p2pToRelaySwitchThreshold=ServerConfig::GetSharedInstance()->GetDouble("p2p_to_relay_switch_threshold", 0.6);
|
|
relayToP2pSwitchThreshold=ServerConfig::GetSharedInstance()->GetDouble("relay_to_p2p_switch_threshold", 0.8);
|
|
reconnectingTimeout=ServerConfig::GetSharedInstance()->GetDouble("reconnecting_state_timeout", 2.0);
|
|
|
|
#ifdef __APPLE__
|
|
machTimestart=0;
|
|
#ifdef TGVOIP_USE_AUDIO_SESSION
|
|
needNotifyAcquiredAudioSession=false;
|
|
#endif
|
|
#endif
|
|
|
|
voip_stream_t* stm=(voip_stream_t *) malloc(sizeof(voip_stream_t));
|
|
stm->id=1;
|
|
stm->type=STREAM_TYPE_AUDIO;
|
|
stm->codec=CODEC_OPUS;
|
|
stm->enabled=1;
|
|
stm->frameDuration=60; //ServerConfig::GetSharedInstance()->GetInt("audio_frame_size", 100);
|
|
outgoingStreams.push_back(stm);
|
|
|
|
memset(signalBarsHistory, 0, sizeof(signalBarsHistory));
|
|
}
|
|
|
|
VoIPController::~VoIPController(){
|
|
LOGD("Entered VoIPController::~VoIPController");
|
|
if(audioInput)
|
|
audioInput->Stop();
|
|
if(audioOutput)
|
|
audioOutput->Stop();
|
|
stopping=true;
|
|
runReceiver=false;
|
|
LOGD("before shutdown socket");
|
|
if(udpSocket)
|
|
udpSocket->Close();
|
|
if(realUdpSocket!=udpSocket)
|
|
realUdpSocket->Close();
|
|
selectCanceller->CancelSelect();
|
|
sendQueue->Put(PendingOutgoingPacket{0});
|
|
if(openingTcpSocket)
|
|
openingTcpSocket->Close();
|
|
|
|
LOGD("before join sendThread");
|
|
join_thread(sendThread);
|
|
LOGD("before join recvThread");
|
|
join_thread(recvThread);
|
|
LOGD("before join tickThread");
|
|
join_thread(tickThread);
|
|
free_mutex(sendBufferMutex);
|
|
LOGD("before close socket");
|
|
if(udpSocket)
|
|
delete udpSocket;
|
|
if(udpSocket!=realUdpSocket)
|
|
delete realUdpSocket;
|
|
LOGD("before delete jitter buffer");
|
|
if(jitterBuffer){
|
|
delete jitterBuffer;
|
|
}
|
|
LOGD("before stop decoder");
|
|
if(decoder){
|
|
decoder->Stop();
|
|
}
|
|
LOGD("before delete audio input");
|
|
if(audioInput){
|
|
delete audioInput;
|
|
}
|
|
LOGD("before delete encoder");
|
|
if(encoder){
|
|
encoder->Stop();
|
|
delete encoder;
|
|
}
|
|
LOGD("before delete audio output");
|
|
if(audioOutput){
|
|
delete audioOutput;
|
|
}
|
|
LOGD("before delete decoder");
|
|
if(decoder){
|
|
delete decoder;
|
|
}
|
|
LOGD("before delete echo canceller");
|
|
if(echoCanceller){
|
|
echoCanceller->Stop();
|
|
delete echoCanceller;
|
|
}
|
|
delete sendQueue;
|
|
unsigned int i;
|
|
for(i=0;i<incomingStreams.size();i++){
|
|
free(incomingStreams[i]);
|
|
}
|
|
incomingStreams.clear();
|
|
for(i=0;i<outgoingStreams.size();i++){
|
|
free(outgoingStreams[i]);
|
|
}
|
|
outgoingStreams.clear();
|
|
free_mutex(queuedPacketsMutex);
|
|
free_mutex(endpointsMutex);
|
|
for(i=0;i<queuedPackets.size();i++){
|
|
if(queuedPackets[i]->data)
|
|
free(queuedPackets[i]->data);
|
|
free(queuedPackets[i]);
|
|
}
|
|
delete conctl;
|
|
for(std::vector<Endpoint*>::iterator itr=endpoints.begin();itr!=endpoints.end();++itr){
|
|
if((*itr)->socket){
|
|
(*itr)->socket->Close();
|
|
delete (*itr)->socket;
|
|
}
|
|
delete *itr;
|
|
}
|
|
if(tgvoipLogFile){
|
|
FILE* log=tgvoipLogFile;
|
|
tgvoipLogFile=NULL;
|
|
fclose(log);
|
|
}
|
|
if(statsDump)
|
|
fclose(statsDump);
|
|
if(resolvedProxyAddress)
|
|
delete resolvedProxyAddress;
|
|
delete selectCanceller;
|
|
if(outputAGC)
|
|
delete outputAGC;
|
|
LOGD("Left VoIPController::~VoIPController");
|
|
}
|
|
|
|
void VoIPController::SetRemoteEndpoints(std::vector<Endpoint> endpoints, bool allowP2p){
|
|
LOGW("Set remote endpoints");
|
|
preferredRelay=NULL;
|
|
size_t i;
|
|
lock_mutex(endpointsMutex);
|
|
this->endpoints.clear();
|
|
didAddTcpRelays=false;
|
|
useTCP=true;
|
|
for(std::vector<Endpoint>::iterator itrtr=endpoints.begin();itrtr!=endpoints.end();++itrtr){
|
|
this->endpoints.push_back(new Endpoint(*itrtr));
|
|
if(itrtr->type==EP_TYPE_TCP_RELAY)
|
|
didAddTcpRelays=true;
|
|
if(itrtr->type==EP_TYPE_UDP_RELAY)
|
|
useTCP=false;
|
|
LOGV("Adding endpoint: %s:%d, %s", itrtr->address.ToString().c_str(), itrtr->port, itrtr->type==EP_TYPE_UDP_RELAY ? "UDP" : "TCP");
|
|
}
|
|
unlock_mutex(endpointsMutex);
|
|
currentEndpoint=this->endpoints[0];
|
|
preferredRelay=currentEndpoint;
|
|
this->allowP2p=allowP2p;
|
|
}
|
|
|
|
void* VoIPController::StartRecvThread(void* controller){
|
|
((VoIPController*)controller)->RunRecvThread();
|
|
return NULL;
|
|
}
|
|
|
|
void* VoIPController::StartSendThread(void* controller){
|
|
((VoIPController*)controller)->RunSendThread();
|
|
return NULL;
|
|
}
|
|
|
|
|
|
void* VoIPController::StartTickThread(void* controller){
|
|
((VoIPController*) controller)->RunTickThread();
|
|
return NULL;
|
|
}
|
|
|
|
|
|
void VoIPController::Start(){
|
|
int res;
|
|
LOGW("========= Starting voip controller =========");
|
|
int32_t cfgFrameSize=60; //ServerConfig::GetSharedInstance()->GetInt("audio_frame_size", 60);
|
|
if(cfgFrameSize==20 || cfgFrameSize==40 || cfgFrameSize==60)
|
|
outgoingStreams[0]->frameDuration=(uint16_t) cfgFrameSize;
|
|
udpSocket->Open();
|
|
if(udpSocket->IsFailed()){
|
|
SetState(STATE_FAILED);
|
|
return;
|
|
}
|
|
|
|
//SendPacket(NULL, 0, currentEndpoint);
|
|
|
|
runReceiver=true;
|
|
start_thread(recvThread, StartRecvThread, this);
|
|
set_thread_priority(recvThread, get_thread_max_priority());
|
|
set_thread_name(recvThread, "voip-recv");
|
|
start_thread(sendThread, StartSendThread, this);
|
|
set_thread_priority(sendThread, get_thread_max_priority());
|
|
set_thread_name(sendThread, "voip-send");
|
|
start_thread(tickThread, StartTickThread, this);
|
|
set_thread_priority(tickThread, get_thread_max_priority());
|
|
set_thread_name(tickThread, "voip-tick");
|
|
}
|
|
|
|
size_t VoIPController::AudioInputCallback(unsigned char* data, size_t length, void* param){
|
|
((VoIPController*)param)->HandleAudioInput(data, length);
|
|
return 0;
|
|
}
|
|
|
|
void VoIPController::HandleAudioInput(unsigned char *data, size_t len){
|
|
if(stopping)
|
|
return;
|
|
if(waitingForAcks || dontSendPackets>0){
|
|
LOGV("waiting for RLC, dropping outgoing audio packet");
|
|
return;
|
|
}
|
|
//LOGE("GOT PACKET OF SIZE %i", len);
|
|
|
|
unsigned char* buf=outgoingPacketsBufferPool.Get();
|
|
if(buf){
|
|
BufferOutputStream pkt(buf, outgoingPacketsBufferPool.GetSingleBufferSize());
|
|
|
|
unsigned char flags=(unsigned char) (len>255 ? STREAM_DATA_FLAG_LEN16 : 0);
|
|
pkt.WriteByte((unsigned char) (1 | flags)); // streamID + flags
|
|
if(len>255)
|
|
pkt.WriteInt16((int16_t) len);
|
|
else
|
|
pkt.WriteByte((unsigned char) len);
|
|
pkt.WriteInt32(audioTimestampOut);
|
|
pkt.WriteBytes(data, len);
|
|
|
|
PendingOutgoingPacket p{
|
|
/*.seq=*/GenerateOutSeq(),
|
|
/*.type=*/PKT_STREAM_DATA,
|
|
/*.len=*/pkt.GetLength(),
|
|
/*.data=*/buf,
|
|
/*.endpoint=*/NULL,
|
|
};
|
|
sendQueue->Put(p);
|
|
} else {
|
|
LOGE("============ COULD NOT GET BUFFER ========");
|
|
}
|
|
|
|
audioTimestampOut+=outgoingStreams[0]->frameDuration;
|
|
}
|
|
|
|
void VoIPController::Connect(){
|
|
assert(state!=STATE_WAIT_INIT_ACK);
|
|
if(proxyProtocol==PROXY_SOCKS5){
|
|
resolvedProxyAddress=NetworkSocket::ResolveDomainName(proxyAddress);
|
|
if(!resolvedProxyAddress){
|
|
LOGW("Error resolving proxy address %s", proxyAddress.c_str());
|
|
SetState(STATE_FAILED);
|
|
return;
|
|
}
|
|
InitUDPProxy();
|
|
}
|
|
connectionInitTime=GetCurrentTime();
|
|
SendInit();
|
|
}
|
|
|
|
|
|
void VoIPController::SetEncryptionKey(char *key, bool isOutgoing){
|
|
memcpy(encryptionKey, key, 256);
|
|
uint8_t sha1[SHA1_LENGTH];
|
|
crypto.sha1((uint8_t*) encryptionKey, 256, sha1);
|
|
memcpy(keyFingerprint, sha1+(SHA1_LENGTH-8), 8);
|
|
uint8_t sha256[SHA256_LENGTH];
|
|
crypto.sha256((uint8_t*) encryptionKey, 256, sha256);
|
|
memcpy(callID, sha256+(SHA256_LENGTH-16), 16);
|
|
this->isOutgoing=isOutgoing;
|
|
}
|
|
|
|
uint32_t VoIPController::GenerateOutSeq(){
|
|
return seq++;
|
|
}
|
|
|
|
void VoIPController::WritePacketHeader(uint32_t pseq, BufferOutputStream *s, unsigned char type, uint32_t length){
|
|
uint32_t acks=0;
|
|
int i;
|
|
for(i=0;i<32;i++){
|
|
if(recvPacketTimes[i]>0)
|
|
acks|=1;
|
|
if(i<31)
|
|
acks<<=1;
|
|
}
|
|
|
|
if(state==STATE_WAIT_INIT || state==STATE_WAIT_INIT_ACK){
|
|
s->WriteInt32(TLID_DECRYPTED_AUDIO_BLOCK);
|
|
int64_t randomID;
|
|
crypto.rand_bytes((uint8_t *) &randomID, 8);
|
|
s->WriteInt64(randomID);
|
|
unsigned char randBytes[7];
|
|
crypto.rand_bytes(randBytes, 7);
|
|
s->WriteByte(7);
|
|
s->WriteBytes(randBytes, 7);
|
|
uint32_t pflags=PFLAG_HAS_RECENT_RECV | PFLAG_HAS_SEQ;
|
|
if(length>0)
|
|
pflags|=PFLAG_HAS_DATA;
|
|
if(state==STATE_WAIT_INIT || state==STATE_WAIT_INIT_ACK){
|
|
pflags|=PFLAG_HAS_CALL_ID | PFLAG_HAS_PROTO;
|
|
}
|
|
pflags|=((uint32_t) type) << 24;
|
|
s->WriteInt32(pflags);
|
|
|
|
if(pflags & PFLAG_HAS_CALL_ID){
|
|
s->WriteBytes(callID, 16);
|
|
}
|
|
s->WriteInt32(lastRemoteSeq);
|
|
s->WriteInt32(pseq);
|
|
s->WriteInt32(acks);
|
|
if(pflags & PFLAG_HAS_PROTO){
|
|
s->WriteInt32(PROTOCOL_NAME);
|
|
}
|
|
if(length>0){
|
|
if(length<=253){
|
|
s->WriteByte((unsigned char) length);
|
|
}else{
|
|
s->WriteByte(254);
|
|
s->WriteByte((unsigned char) (length & 0xFF));
|
|
s->WriteByte((unsigned char) ((length >> 8) & 0xFF));
|
|
s->WriteByte((unsigned char) ((length >> 16) & 0xFF));
|
|
}
|
|
}
|
|
}else{
|
|
s->WriteInt32(TLID_SIMPLE_AUDIO_BLOCK);
|
|
int64_t randomID;
|
|
crypto.rand_bytes((uint8_t *) &randomID, 8);
|
|
s->WriteInt64(randomID);
|
|
unsigned char randBytes[7];
|
|
crypto.rand_bytes(randBytes, 7);
|
|
s->WriteByte(7);
|
|
s->WriteBytes(randBytes, 7);
|
|
uint32_t lenWithHeader=length+13;
|
|
if(lenWithHeader>0){
|
|
if(lenWithHeader<=253){
|
|
s->WriteByte((unsigned char) lenWithHeader);
|
|
}else{
|
|
s->WriteByte(254);
|
|
s->WriteByte((unsigned char) (lenWithHeader & 0xFF));
|
|
s->WriteByte((unsigned char) ((lenWithHeader >> 8) & 0xFF));
|
|
s->WriteByte((unsigned char) ((lenWithHeader >> 16) & 0xFF));
|
|
}
|
|
}
|
|
s->WriteByte(type);
|
|
s->WriteInt32(lastRemoteSeq);
|
|
s->WriteInt32(pseq);
|
|
s->WriteInt32(acks);
|
|
}
|
|
|
|
if(type==PKT_STREAM_DATA || type==PKT_STREAM_DATA_X2 || type==PKT_STREAM_DATA_X3)
|
|
conctl->PacketSent(pseq, length);
|
|
|
|
memmove(&sentPacketTimes[1], sentPacketTimes, 31*sizeof(double));
|
|
sentPacketTimes[0]=GetCurrentTime();
|
|
lastSentSeq=pseq;
|
|
//LOGI("packet header size %d", s->GetLength());
|
|
}
|
|
|
|
|
|
void VoIPController::UpdateAudioBitrate(){
|
|
maxAudioBitrate=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_max_bitrate", 20000);
|
|
maxAudioBitrateGPRS=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_max_bitrate_gprs", 8000);
|
|
maxAudioBitrateEDGE=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_max_bitrate_edge", 16000);
|
|
maxAudioBitrateSaving=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_max_bitrate_saving", 8000);
|
|
initAudioBitrate=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_init_bitrate", 16000);
|
|
initAudioBitrateGPRS=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_init_bitrate_gprs", 8000);
|
|
initAudioBitrateEDGE=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_init_bitrate_edge", 8000);
|
|
initAudioBitrateSaving=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_init_bitrate_saving", 8000);
|
|
audioBitrateStepIncr=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_bitrate_step_incr", 1000);
|
|
audioBitrateStepDecr=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_bitrate_step_decr", 1000);
|
|
minAudioBitrate=(uint32_t) ServerConfig::GetSharedInstance()->GetInt("audio_min_bitrate", 8000);
|
|
relaySwitchThreshold=ServerConfig::GetSharedInstance()->GetDouble("relay_switch_threshold", 0.8);
|
|
p2pToRelaySwitchThreshold=ServerConfig::GetSharedInstance()->GetDouble("p2p_to_relay_switch_threshold", 0.6);
|
|
relayToP2pSwitchThreshold=ServerConfig::GetSharedInstance()->GetDouble("relay_to_p2p_switch_threshold", 0.8);
|
|
reconnectingTimeout=ServerConfig::GetSharedInstance()->GetDouble("reconnecting_state_timeout", 2.0);
|
|
if(encoder){
|
|
if(dataSavingMode || dataSavingRequestedByPeer){
|
|
maxBitrate=maxAudioBitrateSaving;
|
|
encoder->SetBitrate(initAudioBitrateSaving);
|
|
}else if(networkType==NET_TYPE_GPRS){
|
|
maxBitrate=maxAudioBitrateGPRS;
|
|
encoder->SetBitrate(initAudioBitrateGPRS);
|
|
}else if(networkType==NET_TYPE_EDGE){
|
|
maxBitrate=maxAudioBitrateEDGE;
|
|
encoder->SetBitrate(initAudioBitrateEDGE);
|
|
}else{
|
|
maxBitrate=maxAudioBitrate;
|
|
encoder->SetBitrate(initAudioBitrate);
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
void VoIPController::SendInit(){
|
|
lock_mutex(endpointsMutex);
|
|
uint32_t initSeq=GenerateOutSeq();
|
|
for(std::vector<Endpoint*>::iterator itr=endpoints.begin();itr!=endpoints.end();++itr){
|
|
if((*itr)->type==EP_TYPE_TCP_RELAY && !useTCP)
|
|
continue;
|
|
unsigned char* pkt=outgoingPacketsBufferPool.Get();
|
|
if(!pkt){
|
|
LOGW("can't send init, queue overflow");
|
|
continue;
|
|
}
|
|
BufferOutputStream out(pkt, outgoingPacketsBufferPool.GetSingleBufferSize());
|
|
//WritePacketHeader(out, PKT_INIT, 15);
|
|
out.WriteInt32(PROTOCOL_VERSION);
|
|
out.WriteInt32(MIN_PROTOCOL_VERSION);
|
|
uint32_t flags=0;
|
|
if(dataSavingMode)
|
|
flags|=INIT_FLAG_DATA_SAVING_ENABLED;
|
|
out.WriteInt32(flags);
|
|
out.WriteByte(1); // audio codecs count
|
|
out.WriteByte(CODEC_OPUS);
|
|
out.WriteByte(0); // video codecs count
|
|
sendQueue->Put(PendingOutgoingPacket{
|
|
/*.seq=*/initSeq,
|
|
/*.type=*/PKT_INIT,
|
|
/*.len=*/out.GetLength(),
|
|
/*.data=*/pkt,
|
|
/*.endpoint=*/*itr
|
|
});
|
|
}
|
|
unlock_mutex(endpointsMutex);
|
|
SetState(STATE_WAIT_INIT_ACK);
|
|
}
|
|
|
|
void VoIPController::InitUDPProxy(){
|
|
if(realUdpSocket!=udpSocket){
|
|
udpSocket->Close();
|
|
delete udpSocket;
|
|
udpSocket=realUdpSocket;
|
|
}
|
|
NetworkSocket* tcp=NetworkSocket::Create(PROTO_TCP);
|
|
tcp->Connect(resolvedProxyAddress, proxyPort);
|
|
if(tcp->IsFailed()){
|
|
SetState(STATE_FAILED);
|
|
tcp->Close();
|
|
delete tcp;
|
|
return;
|
|
}
|
|
NetworkSocketSOCKS5Proxy* udpProxy=new NetworkSocketSOCKS5Proxy(tcp, udpSocket, proxyUsername, proxyPassword);
|
|
udpProxy->InitConnection();
|
|
udpProxy->Open();
|
|
if(udpProxy->IsFailed()){
|
|
udpProxy->Close();
|
|
delete udpProxy;
|
|
useTCP=true;
|
|
useUDP=false;
|
|
udpConnectivityState=UDP_NOT_AVAILABLE;
|
|
}else{
|
|
udpSocket=udpProxy;
|
|
}
|
|
}
|
|
|
|
void VoIPController::RunRecvThread(){
|
|
LOGI("Receive thread starting");
|
|
unsigned char buffer[1024];
|
|
NetworkPacket packet;
|
|
while(runReceiver){
|
|
packet.data=buffer;
|
|
packet.length=1024;
|
|
|
|
std::vector<NetworkSocket*> readSockets;
|
|
std::vector<NetworkSocket*> errorSockets;
|
|
readSockets.push_back(realUdpSocket);
|
|
errorSockets.push_back(realUdpSocket);
|
|
|
|
for(std::vector<Endpoint*>::iterator itr=endpoints.begin();itr!=endpoints.end();++itr){
|
|
if((*itr)->type==EP_TYPE_TCP_RELAY){
|
|
if((*itr)->socket){
|
|
readSockets.push_back((*itr)->socket);
|
|
errorSockets.push_back((*itr)->socket);
|
|
}
|
|
}
|
|
}
|
|
|
|
bool selRes=NetworkSocket::Select(readSockets, errorSockets, selectCanceller);
|
|
if(!selRes){
|
|
LOGV("Select canceled");
|
|
continue;
|
|
}
|
|
if(!runReceiver)
|
|
return;
|
|
|
|
if(!errorSockets.empty()){
|
|
if(std::find(errorSockets.begin(), errorSockets.end(), realUdpSocket)!=errorSockets.end()){
|
|
LOGW("UDP socket failed");
|
|
SetState(STATE_FAILED);
|
|
return;
|
|
}
|
|
}
|
|
|
|
NetworkSocket* socket=NULL;
|
|
|
|
if(std::find(readSockets.begin(), readSockets.end(), realUdpSocket)!=readSockets.end()){
|
|
socket=udpSocket;
|
|
}else if(readSockets.size()>0){
|
|
socket=readSockets[0];
|
|
}else{
|
|
LOGI("no sockets to read from");
|
|
lock_mutex(endpointsMutex);
|
|
for(std::vector<NetworkSocket*>::iterator itr=errorSockets.begin();itr!=errorSockets.end();++itr){
|
|
for(std::vector<Endpoint*>::iterator e=endpoints.begin();e!=endpoints.end();++e){
|
|
if((*e)->socket && (*e)->socket==*itr){
|
|
(*e)->socket->Close();
|
|
delete (*e)->socket;
|
|
(*e)->socket=NULL;
|
|
LOGI("Closing failed TCP socket for %s:%u", (*e)->address.ToString().c_str(), (*e)->port);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
unlock_mutex(endpointsMutex);
|
|
continue;
|
|
}
|
|
|
|
socket->Receive(&packet);
|
|
if(!packet.address){
|
|
LOGE("Packet has null address. This shouldn't happen.");
|
|
continue;
|
|
}
|
|
size_t len=packet.length;
|
|
if(!len){
|
|
LOGE("Packet has zero length.");
|
|
continue;
|
|
}
|
|
//LOGV("Received %d bytes from %s:%d at %.5lf", len, packet.address->ToString().c_str(), packet.port, GetCurrentTime());
|
|
Endpoint* srcEndpoint=NULL;
|
|
|
|
IPv4Address* src4=dynamic_cast<IPv4Address*>(packet.address);
|
|
if(src4){
|
|
lock_mutex(endpointsMutex);
|
|
for(std::vector<Endpoint*>::iterator itrtr=endpoints.begin();itrtr!=endpoints.end();++itrtr){
|
|
if((*itrtr)->address==*src4 && (*itrtr)->port==packet.port){
|
|
if(((*itrtr)->type!=EP_TYPE_TCP_RELAY && packet.protocol==PROTO_UDP) || ((*itrtr)->type==EP_TYPE_TCP_RELAY && packet.protocol==PROTO_TCP)){
|
|
srcEndpoint=*itrtr;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
unlock_mutex(endpointsMutex);
|
|
}
|
|
|
|
if(!srcEndpoint){
|
|
LOGW("Received a packet from unknown source %s:%u", packet.address->ToString().c_str(), packet.port);
|
|
continue;
|
|
}
|
|
if(len<=0){
|
|
//LOGW("error receiving: %d / %s", errno, strerror(errno));
|
|
continue;
|
|
}
|
|
if(IS_MOBILE_NETWORK(networkType))
|
|
stats.bytesRecvdMobile+=(uint64_t)len;
|
|
else
|
|
stats.bytesRecvdWifi+=(uint64_t)len;
|
|
try{
|
|
if(memcmp(buffer, srcEndpoint->type==EP_TYPE_UDP_RELAY || srcEndpoint->type==EP_TYPE_TCP_RELAY ? (void*)srcEndpoint->peerTag : (void*)callID, 16)!=0){
|
|
LOGW("Received packet has wrong peerTag");
|
|
|
|
continue;
|
|
}
|
|
BufferInputStream in(buffer, (size_t)len);
|
|
in.Seek(16);
|
|
if(in.Remaining()>=16 && (srcEndpoint->type==EP_TYPE_UDP_RELAY || srcEndpoint->type==EP_TYPE_TCP_RELAY)
|
|
&& *reinterpret_cast<uint64_t*>(buffer+16)==0xFFFFFFFFFFFFFFFFLL && *reinterpret_cast<uint32_t*>(buffer+24)==0xFFFFFFFF){
|
|
// relay special request response
|
|
in.Seek(16+12);
|
|
uint32_t tlid=(uint32_t) in.ReadInt32();
|
|
|
|
if(tlid==TLID_UDP_REFLECTOR_SELF_INFO){
|
|
if(srcEndpoint->type==EP_TYPE_UDP_RELAY && udpConnectivityState==UDP_PING_SENT && in.Remaining()>=32){
|
|
int32_t date=in.ReadInt32();
|
|
int64_t queryID=in.ReadInt64();
|
|
unsigned char myIP[16];
|
|
in.ReadBytes(myIP, 16);
|
|
int32_t myPort=in.ReadInt32();
|
|
udpConnectivityState=UDP_AVAILABLE;
|
|
//LOGV("Received UDP ping reply from %s:%d: date=%d, queryID=%lld, my IP=%s, my port=%d", srcEndpoint->address.ToString().c_str(), srcEndpoint->port, date, queryID, IPv6Address(myIP).ToString().c_str(), myPort);
|
|
}
|
|
}else if(tlid==TLID_UDP_REFLECTOR_PEER_INFO){
|
|
if(waitingForRelayPeerInfo && in.Remaining()>=16){
|
|
lock_mutex(endpointsMutex);
|
|
uint32_t myAddr=(uint32_t) in.ReadInt32();
|
|
uint32_t myPort=(uint32_t) in.ReadInt32();
|
|
uint32_t peerAddr=(uint32_t) in.ReadInt32();
|
|
uint32_t peerPort=(uint32_t) in.ReadInt32();
|
|
for(std::vector<Endpoint *>::iterator itrtr=endpoints.begin(); itrtr!=endpoints.end(); ++itrtr){
|
|
Endpoint *ep=*itrtr;
|
|
if(ep->type==EP_TYPE_UDP_P2P_INET){
|
|
if(currentEndpoint==ep)
|
|
currentEndpoint=preferredRelay;
|
|
delete ep;
|
|
endpoints.erase(itrtr);
|
|
break;
|
|
}
|
|
}
|
|
for(std::vector<Endpoint *>::iterator itrtr=endpoints.begin(); itrtr!=endpoints.end(); ++itrtr){
|
|
Endpoint *ep=*itrtr;
|
|
if(ep->type==EP_TYPE_UDP_P2P_LAN){
|
|
if(currentEndpoint==ep)
|
|
currentEndpoint=preferredRelay;
|
|
delete ep;
|
|
endpoints.erase(itrtr);
|
|
break;
|
|
}
|
|
}
|
|
IPv4Address _peerAddr(peerAddr);
|
|
IPv6Address emptyV6("::0");
|
|
unsigned char peerTag[16];
|
|
endpoints.push_back(new Endpoint(0, (uint16_t) peerPort, _peerAddr, emptyV6, EP_TYPE_UDP_P2P_INET, peerTag));
|
|
LOGW("Received reflector peer info, my=%08X:%u, peer=%08X:%u", myAddr, myPort, peerAddr, peerPort);
|
|
if(myAddr==peerAddr){
|
|
LOGW("Detected LAN");
|
|
IPv4Address lanAddr(0);
|
|
udpSocket->GetLocalInterfaceInfo(&lanAddr, NULL);
|
|
|
|
BufferOutputStream pkt(8);
|
|
pkt.WriteInt32(lanAddr.GetAddress());
|
|
pkt.WriteInt32(udpSocket->GetLocalPort());
|
|
SendPacketReliably(PKT_LAN_ENDPOINT, pkt.GetBuffer(), pkt.GetLength(), 0.5, 10);
|
|
}
|
|
unlock_mutex(endpointsMutex);
|
|
waitingForRelayPeerInfo=false;
|
|
}
|
|
}else{
|
|
LOGV("Received relay response with unknown tl id: 0x%08X", tlid);
|
|
}
|
|
continue;
|
|
}
|
|
if(in.Remaining()<40){
|
|
LOGV("Received packet is too small");
|
|
continue;
|
|
}
|
|
|
|
unsigned char fingerprint[8], msgHash[16];
|
|
in.ReadBytes(fingerprint, 8);
|
|
in.ReadBytes(msgHash, 16);
|
|
if(memcmp(fingerprint, keyFingerprint, 8)!=0){
|
|
LOGW("Received packet has wrong key fingerprint");
|
|
|
|
continue;
|
|
}
|
|
unsigned char key[32], iv[32];
|
|
KDF(msgHash, isOutgoing ? 8 : 0, key, iv);
|
|
unsigned char aesOut[MSC_STACK_FALLBACK(in.Remaining(), 1024)];
|
|
crypto.aes_ige_decrypt((unsigned char *) buffer+in.GetOffset(), aesOut, in.Remaining(), key, iv);
|
|
memcpy(buffer+in.GetOffset(), aesOut, in.Remaining());
|
|
unsigned char sha[SHA1_LENGTH];
|
|
uint32_t _len=(uint32_t) in.ReadInt32();
|
|
if(_len>in.Remaining())
|
|
_len=in.Remaining();
|
|
crypto.sha1((uint8_t *) (buffer+in.GetOffset()-4), (size_t) (_len+4), sha);
|
|
if(memcmp(msgHash, sha+(SHA1_LENGTH-16), 16)!=0){
|
|
LOGW("Received packet has wrong hash after decryption");
|
|
|
|
continue;
|
|
}
|
|
|
|
lastRecvPacketTime=GetCurrentTime();
|
|
|
|
|
|
/*decryptedAudioBlock random_id:long random_bytes:string flags:# voice_call_id:flags.2?int128 in_seq_no:flags.4?int out_seq_no:flags.4?int
|
|
* recent_received_mask:flags.5?int proto:flags.3?int extra:flags.1?string raw_data:flags.0?string = DecryptedAudioBlock
|
|
simpleAudioBlock random_id:long random_bytes:string raw_data:string = DecryptedAudioBlock;
|
|
*/
|
|
uint32_t ackId, pseq, acks;
|
|
unsigned char type;
|
|
uint32_t tlid=(uint32_t) in.ReadInt32();
|
|
uint32_t packetInnerLen;
|
|
if(tlid==TLID_DECRYPTED_AUDIO_BLOCK){
|
|
in.ReadInt64(); // random id
|
|
uint32_t randLen=(uint32_t) in.ReadTlLength();
|
|
in.Seek(in.GetOffset()+randLen+pad4(randLen));
|
|
uint32_t flags=(uint32_t) in.ReadInt32();
|
|
type=(unsigned char) ((flags >> 24) & 0xFF);
|
|
if(!(flags & PFLAG_HAS_SEQ && flags & PFLAG_HAS_RECENT_RECV)){
|
|
LOGW("Received packet doesn't have PFLAG_HAS_SEQ, PFLAG_HAS_RECENT_RECV, or both");
|
|
|
|
continue;
|
|
}
|
|
if(flags & PFLAG_HAS_CALL_ID){
|
|
unsigned char pktCallID[16];
|
|
in.ReadBytes(pktCallID, 16);
|
|
if(memcmp(pktCallID, callID, 16)!=0){
|
|
LOGW("Received packet has wrong call id");
|
|
|
|
lastError=TGVOIP_ERROR_UNKNOWN;
|
|
SetState(STATE_FAILED);
|
|
return;
|
|
}
|
|
}
|
|
ackId=(uint32_t) in.ReadInt32();
|
|
pseq=(uint32_t) in.ReadInt32();
|
|
acks=(uint32_t) in.ReadInt32();
|
|
if(flags & PFLAG_HAS_PROTO){
|
|
uint32_t proto=(uint32_t) in.ReadInt32();
|
|
if(proto!=PROTOCOL_NAME){
|
|
LOGW("Received packet uses wrong protocol");
|
|
|
|
lastError=TGVOIP_ERROR_INCOMPATIBLE;
|
|
SetState(STATE_FAILED);
|
|
return;
|
|
}
|
|
}
|
|
if(flags & PFLAG_HAS_EXTRA){
|
|
uint32_t extraLen=(uint32_t) in.ReadTlLength();
|
|
in.Seek(in.GetOffset()+extraLen+pad4(extraLen));
|
|
}
|
|
if(flags & PFLAG_HAS_DATA){
|
|
packetInnerLen=in.ReadTlLength();
|
|
}
|
|
}else if(tlid==TLID_SIMPLE_AUDIO_BLOCK){
|
|
in.ReadInt64(); // random id
|
|
uint32_t randLen=(uint32_t) in.ReadTlLength();
|
|
in.Seek(in.GetOffset()+randLen+pad4(randLen));
|
|
packetInnerLen=in.ReadTlLength();
|
|
type=in.ReadByte();
|
|
ackId=(uint32_t) in.ReadInt32();
|
|
pseq=(uint32_t) in.ReadInt32();
|
|
acks=(uint32_t) in.ReadInt32();
|
|
}else{
|
|
LOGW("Received a packet of unknown type %08X", tlid);
|
|
|
|
continue;
|
|
}
|
|
packetsRecieved++;
|
|
if(seqgt(pseq, lastRemoteSeq)){
|
|
uint32_t diff=pseq-lastRemoteSeq;
|
|
if(diff>31){
|
|
memset(recvPacketTimes, 0, 32*sizeof(double));
|
|
}else{
|
|
memmove(&recvPacketTimes[diff], recvPacketTimes, (32-diff)*sizeof(double));
|
|
if(diff>1){
|
|
memset(recvPacketTimes, 0, diff*sizeof(double));
|
|
}
|
|
recvPacketTimes[0]=GetCurrentTime();
|
|
}
|
|
lastRemoteSeq=pseq;
|
|
}else if(!seqgt(pseq, lastRemoteSeq) && lastRemoteSeq-pseq<32){
|
|
if(recvPacketTimes[lastRemoteSeq-pseq]!=0){
|
|
LOGW("Received duplicated packet for seq %u", pseq);
|
|
|
|
continue;
|
|
}
|
|
recvPacketTimes[lastRemoteSeq-pseq]=GetCurrentTime();
|
|
}else if(lastRemoteSeq-pseq>=32){
|
|
LOGW("Packet %u is out of order and too late", pseq);
|
|
|
|
continue;
|
|
}
|
|
if(seqgt(ackId, lastRemoteAckSeq)){
|
|
uint32_t diff=ackId-lastRemoteAckSeq;
|
|
if(diff>31){
|
|
memset(remoteAcks, 0, 32*sizeof(double));
|
|
}else{
|
|
memmove(&remoteAcks[diff], remoteAcks, (32-diff)*sizeof(double));
|
|
if(diff>1){
|
|
memset(remoteAcks, 0, diff*sizeof(double));
|
|
}
|
|
remoteAcks[0]=GetCurrentTime();
|
|
}
|
|
if(waitingForAcks && lastRemoteAckSeq>=firstSentPing){
|
|
memset(rttHistory, 0, 32*sizeof(double));
|
|
waitingForAcks=false;
|
|
dontSendPackets=10;
|
|
LOGI("resuming sending");
|
|
}
|
|
lastRemoteAckSeq=ackId;
|
|
conctl->PacketAcknowledged(ackId);
|
|
int i;
|
|
for(i=0;i<31;i++){
|
|
if(remoteAcks[i+1]==0){
|
|
if((acks >> (31-i)) & 1){
|
|
remoteAcks[i+1]=GetCurrentTime();
|
|
conctl->PacketAcknowledged(ackId-(i+1));
|
|
}
|
|
}
|
|
}
|
|
lock_mutex(queuedPacketsMutex);
|
|
for(i=0;i<queuedPackets.size();i++){
|
|
voip_queued_packet_t* qp=queuedPackets[i];
|
|
int j;
|
|
bool didAck=false;
|
|
for(j=0;j<16;j++){
|
|
LOGD("queued packet %u, seq %u=%u", i, j, qp->seqs[j]);
|
|
if(qp->seqs[j]==0)
|
|
break;
|
|
int remoteAcksIndex=lastRemoteAckSeq-qp->seqs[j];
|
|
LOGV("remote acks index %u, value %f", remoteAcksIndex, remoteAcksIndex>=0 && remoteAcksIndex<32 ? remoteAcks[remoteAcksIndex] : -1);
|
|
if(seqgt(lastRemoteAckSeq, qp->seqs[j]) && remoteAcksIndex>=0 && remoteAcksIndex<32 && remoteAcks[remoteAcksIndex]>0){
|
|
LOGD("did ack seq %u, removing", qp->seqs[j]);
|
|
didAck=true;
|
|
break;
|
|
}
|
|
}
|
|
if(didAck){
|
|
if(qp->data)
|
|
free(qp->data);
|
|
free(qp);
|
|
queuedPackets.erase(queuedPackets.begin()+i);
|
|
i--;
|
|
continue;
|
|
}
|
|
}
|
|
unlock_mutex(queuedPacketsMutex);
|
|
}
|
|
|
|
if(srcEndpoint!=currentEndpoint && (srcEndpoint->type==EP_TYPE_UDP_RELAY || srcEndpoint->type==EP_TYPE_TCP_RELAY) && ((currentEndpoint->type!=EP_TYPE_UDP_RELAY && currentEndpoint->type!=EP_TYPE_TCP_RELAY) || currentEndpoint->averageRTT==0)){
|
|
if(seqgt(lastSentSeq-32, lastRemoteAckSeq)){
|
|
currentEndpoint=srcEndpoint;
|
|
LOGI("Peer network address probably changed, switching to relay");
|
|
if(allowP2p)
|
|
SendPublicEndpointsRequest();
|
|
}
|
|
}
|
|
//LOGV("acks: %u -> %.2lf, %.2lf, %.2lf, %.2lf, %.2lf, %.2lf, %.2lf, %.2lf", lastRemoteAckSeq, remoteAcks[0], remoteAcks[1], remoteAcks[2], remoteAcks[3], remoteAcks[4], remoteAcks[5], remoteAcks[6], remoteAcks[7]);
|
|
//LOGD("recv: %u -> %.2lf, %.2lf, %.2lf, %.2lf, %.2lf, %.2lf, %.2lf, %.2lf", lastRemoteSeq, recvPacketTimes[0], recvPacketTimes[1], recvPacketTimes[2], recvPacketTimes[3], recvPacketTimes[4], recvPacketTimes[5], recvPacketTimes[6], recvPacketTimes[7]);
|
|
//LOGI("RTT = %.3lf", GetAverageRTT());
|
|
//LOGV("Packet %u type is %d", pseq, type);
|
|
if(type==PKT_INIT){
|
|
LOGD("Received init");
|
|
if(!receivedInit){
|
|
receivedInit=true;
|
|
currentEndpoint=srcEndpoint;
|
|
if(srcEndpoint->type==EP_TYPE_UDP_RELAY || (useTCP && srcEndpoint->type==EP_TYPE_TCP_RELAY))
|
|
preferredRelay=srcEndpoint;
|
|
LogDebugInfo();
|
|
}
|
|
peerVersion=(uint32_t) in.ReadInt32();
|
|
LOGI("Peer version is %d", peerVersion);
|
|
uint32_t minVer=(uint32_t) in.ReadInt32();
|
|
if(minVer>PROTOCOL_VERSION || peerVersion<MIN_PROTOCOL_VERSION){
|
|
lastError=TGVOIP_ERROR_INCOMPATIBLE;
|
|
|
|
SetState(STATE_FAILED);
|
|
return;
|
|
}
|
|
uint32_t flags=(uint32_t) in.ReadInt32();
|
|
if(flags & INIT_FLAG_DATA_SAVING_ENABLED){
|
|
dataSavingRequestedByPeer=true;
|
|
UpdateDataSavingState();
|
|
UpdateAudioBitrate();
|
|
}
|
|
|
|
int i;
|
|
int numSupportedAudioCodecs=in.ReadByte();
|
|
for(i=0; i<numSupportedAudioCodecs; i++){
|
|
in.ReadByte(); // ignore for now
|
|
}
|
|
int numSupportedVideoCodecs=in.ReadByte();
|
|
for(i=0; i<numSupportedVideoCodecs; i++){
|
|
in.ReadByte(); // ignore for now
|
|
}
|
|
|
|
unsigned char* buf=outgoingPacketsBufferPool.Get();
|
|
if(buf){
|
|
BufferOutputStream out(buf, outgoingPacketsBufferPool.GetSingleBufferSize());
|
|
//WritePacketHeader(out, PKT_INIT_ACK, (peerVersion>=2 ? 10 : 2)+(peerVersion>=2 ? 6 : 4)*outgoingStreams.size());
|
|
|
|
out.WriteInt32(PROTOCOL_VERSION);
|
|
out.WriteInt32(MIN_PROTOCOL_VERSION);
|
|
|
|
out.WriteByte((unsigned char) outgoingStreams.size());
|
|
for(i=0; i<outgoingStreams.size(); i++){
|
|
out.WriteByte(outgoingStreams[i]->id);
|
|
out.WriteByte(outgoingStreams[i]->type);
|
|
out.WriteByte(outgoingStreams[i]->codec);
|
|
out.WriteInt16(outgoingStreams[i]->frameDuration);
|
|
out.WriteByte((unsigned char) (outgoingStreams[i]->enabled ? 1 : 0));
|
|
}
|
|
sendQueue->Put(PendingOutgoingPacket{
|
|
/*.seq=*/GenerateOutSeq(),
|
|
/*.type=*/PKT_INIT_ACK,
|
|
/*.len=*/out.GetLength(),
|
|
/*.data=*/buf,
|
|
/*.endpoint=*/NULL
|
|
});
|
|
}
|
|
}
|
|
if(type==PKT_INIT_ACK){
|
|
LOGD("Received init ack");
|
|
|
|
if(!receivedInitAck){
|
|
receivedInitAck=true;
|
|
if(packetInnerLen>10){
|
|
peerVersion=in.ReadInt32();
|
|
uint32_t minVer=(uint32_t) in.ReadInt32();
|
|
if(minVer>PROTOCOL_VERSION || peerVersion<MIN_PROTOCOL_VERSION){
|
|
lastError=TGVOIP_ERROR_INCOMPATIBLE;
|
|
|
|
SetState(STATE_FAILED);
|
|
return;
|
|
}
|
|
}else{
|
|
peerVersion=1;
|
|
}
|
|
|
|
LOGI("peer version from init ack %d", peerVersion);
|
|
|
|
unsigned char streamCount=in.ReadByte();
|
|
if(streamCount==0)
|
|
continue;
|
|
|
|
int i;
|
|
voip_stream_t *incomingAudioStream=NULL;
|
|
for(i=0; i<streamCount; i++){
|
|
voip_stream_t *stm=(voip_stream_t *) malloc(sizeof(voip_stream_t));
|
|
stm->id=in.ReadByte();
|
|
stm->type=in.ReadByte();
|
|
stm->codec=in.ReadByte();
|
|
if(peerVersion>=2)
|
|
stm->frameDuration=(uint16_t) in.ReadInt16();
|
|
else
|
|
stm->frameDuration=20;
|
|
stm->enabled=in.ReadByte()==1;
|
|
incomingStreams.push_back(stm);
|
|
if(stm->type==STREAM_TYPE_AUDIO && !incomingAudioStream)
|
|
incomingAudioStream=stm;
|
|
}
|
|
if(!incomingAudioStream)
|
|
continue;
|
|
|
|
voip_stream_t *outgoingAudioStream=outgoingStreams[0];
|
|
|
|
if(!audioInput){
|
|
LOGI("before create audio io");
|
|
audioInput=tgvoip::audio::AudioInput::Create(currentAudioInput, this);
|
|
audioInput->Configure(48000, 16, 1);
|
|
audioOutput=tgvoip::audio::AudioOutput::Create(currentAudioOutput, this);
|
|
audioOutput->Configure(48000, 16, 1);
|
|
echoCanceller=new EchoCanceller(config.enableAEC, config.enableNS, config.enableAGC);
|
|
echoCanceller->SetAECStrength(echoCancellationStrength);
|
|
encoder=new OpusEncoder(audioInput);
|
|
encoder->SetCallback(AudioInputCallback, this);
|
|
encoder->SetOutputFrameDuration(outgoingAudioStream->frameDuration);
|
|
encoder->SetEchoCanceller(echoCanceller);
|
|
encoder->Start();
|
|
if(!micMuted){
|
|
audioInput->Start();
|
|
if(!audioInput->IsInitialized()){
|
|
LOGE("Erorr initializing audio capture");
|
|
lastError=TGVOIP_ERROR_AUDIO_IO;
|
|
|
|
SetState(STATE_FAILED);
|
|
return;
|
|
}
|
|
}
|
|
if(!audioOutput->IsInitialized()){
|
|
LOGE("Erorr initializing audio playback");
|
|
lastError=TGVOIP_ERROR_AUDIO_IO;
|
|
|
|
SetState(STATE_FAILED);
|
|
return;
|
|
}
|
|
UpdateAudioBitrate();
|
|
|
|
jitterBuffer=new JitterBuffer(NULL, incomingAudioStream->frameDuration);
|
|
outputAGC=new AutomaticGainControl();
|
|
outputAGC->SetPassThrough(!outputAGCEnabled);
|
|
decoder=new OpusDecoder(audioOutput);
|
|
decoder->SetEchoCanceller(echoCanceller);
|
|
decoder->SetJitterBuffer(jitterBuffer);
|
|
decoder->SetFrameDuration(incomingAudioStream->frameDuration);
|
|
decoder->AddAudioEffect(outputAGC);
|
|
decoder->Start();
|
|
if(incomingAudioStream->frameDuration>50)
|
|
jitterBuffer->SetMinPacketCount((uint32_t) ServerConfig::GetSharedInstance()->GetInt("jitter_initial_delay_60", 3));
|
|
else if(incomingAudioStream->frameDuration>30)
|
|
jitterBuffer->SetMinPacketCount((uint32_t) ServerConfig::GetSharedInstance()->GetInt("jitter_initial_delay_40", 4));
|
|
else
|
|
jitterBuffer->SetMinPacketCount((uint32_t) ServerConfig::GetSharedInstance()->GetInt("jitter_initial_delay_20", 6));
|
|
//audioOutput->Start();
|
|
#ifdef TGVOIP_USE_AUDIO_SESSION
|
|
#ifdef __APPLE__
|
|
if(acquireAudioSession){
|
|
acquireAudioSession(^(){
|
|
LOGD("Audio session acquired");
|
|
needNotifyAcquiredAudioSession=true;
|
|
});
|
|
}else{
|
|
audio::AudioUnitIO::AudioSessionAcquired();
|
|
}
|
|
#endif
|
|
#endif
|
|
}
|
|
setEstablishedAt=GetCurrentTime()+ServerConfig::GetSharedInstance()->GetDouble("established_delay_if_no_stream_data", 1.5);
|
|
if(allowP2p)
|
|
SendPublicEndpointsRequest();
|
|
}
|
|
}
|
|
if(type==PKT_STREAM_DATA || type==PKT_STREAM_DATA_X2 || type==PKT_STREAM_DATA_X3){
|
|
if(state!=STATE_ESTABLISHED && receivedInitAck)
|
|
SetState(STATE_ESTABLISHED);
|
|
int count;
|
|
switch(type){
|
|
case PKT_STREAM_DATA_X2:
|
|
count=2;
|
|
break;
|
|
case PKT_STREAM_DATA_X3:
|
|
count=3;
|
|
break;
|
|
case PKT_STREAM_DATA:
|
|
default:
|
|
count=1;
|
|
break;
|
|
}
|
|
int i;
|
|
if(srcEndpoint->type==EP_TYPE_UDP_RELAY && srcEndpoint!=peerPreferredRelay){
|
|
peerPreferredRelay=srcEndpoint;
|
|
}
|
|
for(i=0;i<count;i++){
|
|
unsigned char streamID=in.ReadByte();
|
|
unsigned char flags=(unsigned char) (streamID & 0xC0);
|
|
uint16_t sdlen=(uint16_t) (flags & STREAM_DATA_FLAG_LEN16 ? in.ReadInt16() : in.ReadByte());
|
|
uint32_t pts=(uint32_t) in.ReadInt32();
|
|
//LOGD("stream data, pts=%d, len=%d, rem=%d", pts, sdlen, in.Remaining());
|
|
audioTimestampIn=pts;
|
|
if(!audioOutStarted && audioOutput){
|
|
audioOutput->Start();
|
|
audioOutStarted=true;
|
|
}
|
|
if(jitterBuffer && in.Remaining()>=sdlen)
|
|
jitterBuffer->HandleInput((unsigned char*) (buffer+in.GetOffset()), sdlen, pts);
|
|
if(i<count-1)
|
|
in.Seek(in.GetOffset()+sdlen);
|
|
}
|
|
}
|
|
if(type==PKT_PING){
|
|
LOGD("Received ping from %s:%d", packet.address->ToString().c_str(), srcEndpoint->port);
|
|
if(srcEndpoint->type!=EP_TYPE_UDP_RELAY && srcEndpoint->type!=EP_TYPE_TCP_RELAY && !allowP2p){
|
|
LOGW("Received p2p ping but p2p is disabled by manual override");
|
|
continue;
|
|
}
|
|
unsigned char* buf=outgoingPacketsBufferPool.Get();
|
|
if(!buf){
|
|
LOGW("Dropping pong packet, queue overflow");
|
|
continue;
|
|
}
|
|
BufferOutputStream pkt(buf, outgoingPacketsBufferPool.GetSingleBufferSize());
|
|
pkt.WriteInt32(pseq);
|
|
sendQueue->Put(PendingOutgoingPacket{
|
|
/*.seq=*/GenerateOutSeq(),
|
|
/*.type=*/PKT_PONG,
|
|
/*.len=*/pkt.GetLength(),
|
|
/*.data=*/buf,
|
|
/*.endpoint=*/srcEndpoint,
|
|
});
|
|
}
|
|
if(type==PKT_PONG){
|
|
if(packetInnerLen>=4){
|
|
uint32_t pingSeq=(uint32_t) in.ReadInt32();
|
|
if(pingSeq==srcEndpoint->lastPingSeq){
|
|
memmove(&srcEndpoint->rtts[1], srcEndpoint->rtts, sizeof(double)*5);
|
|
srcEndpoint->rtts[0]=GetCurrentTime()-srcEndpoint->lastPingTime;
|
|
int i;
|
|
srcEndpoint->averageRTT=0;
|
|
for(i=0;i<6;i++){
|
|
if(srcEndpoint->rtts[i]==0)
|
|
break;
|
|
srcEndpoint->averageRTT+=srcEndpoint->rtts[i];
|
|
}
|
|
srcEndpoint->averageRTT/=i;
|
|
LOGD("Current RTT via %s: %.3f, average: %.3f", packet.address->ToString().c_str(), srcEndpoint->rtts[0], srcEndpoint->averageRTT);
|
|
}
|
|
}
|
|
/*if(currentEndpoint!=srcEndpoint && (srcEndpoint->type==EP_TYPE_UDP_P2P_INET || srcEndpoint->type==EP_TYPE_UDP_P2P_LAN)){
|
|
LOGI("Switching to P2P now!");
|
|
currentEndpoint=srcEndpoint;
|
|
needSendP2pPing=false;
|
|
}*/
|
|
}
|
|
if(type==PKT_STREAM_STATE){
|
|
unsigned char id=in.ReadByte();
|
|
unsigned char enabled=in.ReadByte();
|
|
int i;
|
|
for(i=0;i<incomingStreams.size();i++){
|
|
if(incomingStreams[i]->id==id){
|
|
incomingStreams[i]->enabled=enabled==1;
|
|
UpdateAudioOutputState();
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
if(type==PKT_LAN_ENDPOINT){
|
|
LOGV("received lan endpoint");
|
|
uint32_t peerAddr=(uint32_t) in.ReadInt32();
|
|
uint16_t peerPort=(uint16_t) in.ReadInt32();
|
|
lock_mutex(endpointsMutex);
|
|
for(std::vector<Endpoint*>::iterator itrtr=endpoints.begin();itrtr!=endpoints.end();++itrtr){
|
|
if((*itrtr)->type==EP_TYPE_UDP_P2P_LAN){
|
|
if(currentEndpoint==*itrtr)
|
|
currentEndpoint=preferredRelay;
|
|
delete *itrtr;
|
|
endpoints.erase(itrtr);
|
|
break;
|
|
}
|
|
}
|
|
IPv4Address v4addr(peerAddr);
|
|
IPv6Address v6addr("::0");
|
|
unsigned char peerTag[16];
|
|
endpoints.push_back(new Endpoint(0, peerPort, v4addr, v6addr, EP_TYPE_UDP_P2P_LAN, peerTag));
|
|
unlock_mutex(endpointsMutex);
|
|
}
|
|
if(type==PKT_NETWORK_CHANGED && currentEndpoint->type!=EP_TYPE_UDP_RELAY && currentEndpoint->type!=EP_TYPE_TCP_RELAY){
|
|
currentEndpoint=preferredRelay;
|
|
if(allowP2p)
|
|
SendPublicEndpointsRequest();
|
|
if(peerVersion>=2){
|
|
uint32_t flags=(uint32_t) in.ReadInt32();
|
|
dataSavingRequestedByPeer=(flags & INIT_FLAG_DATA_SAVING_ENABLED)==INIT_FLAG_DATA_SAVING_ENABLED;
|
|
UpdateDataSavingState();
|
|
UpdateAudioBitrate();
|
|
}
|
|
}
|
|
}catch(std::out_of_range x){
|
|
LOGW("Error parsing packet: %s", x.what());
|
|
}
|
|
}
|
|
LOGI("=== recv thread exiting ===");
|
|
}
|
|
|
|
void VoIPController::RunSendThread(){
|
|
unsigned char buf[1500];
|
|
while(runReceiver){
|
|
PendingOutgoingPacket pkt=sendQueue->GetBlocking();
|
|
if(pkt.data){
|
|
lock_mutex(endpointsMutex);
|
|
Endpoint *endpoint=pkt.endpoint ? pkt.endpoint : currentEndpoint;
|
|
if((endpoint->type==EP_TYPE_TCP_RELAY && useTCP) || (endpoint->type!=EP_TYPE_TCP_RELAY && useUDP)){
|
|
BufferOutputStream p(buf, sizeof(buf));
|
|
WritePacketHeader(pkt.seq, &p, pkt.type, pkt.len);
|
|
p.WriteBytes(pkt.data, pkt.len);
|
|
SendPacket(p.GetBuffer(), p.GetLength(), endpoint);
|
|
}
|
|
unlock_mutex(endpointsMutex);
|
|
outgoingPacketsBufferPool.Reuse(pkt.data);
|
|
}else{
|
|
LOGE("tried to send null packet");
|
|
}
|
|
}
|
|
LOGI("=== send thread exiting ===");
|
|
}
|
|
|
|
|
|
void VoIPController::RunTickThread(){
|
|
uint32_t tickCount=0;
|
|
bool wasWaitingForAcks=false;
|
|
double startTime=GetCurrentTime();
|
|
while(runReceiver){
|
|
#ifndef _WIN32
|
|
usleep(100000);
|
|
#else
|
|
Sleep(100);
|
|
#endif
|
|
int prevSignalBarCount=GetSignalBarsCount();
|
|
signalBarCount=4;
|
|
tickCount++;
|
|
if(connectionInitTime==0)
|
|
continue;
|
|
double time=GetCurrentTime();
|
|
if(state==STATE_RECONNECTING)
|
|
signalBarCount=1;
|
|
if(tickCount%5==0 && (state==STATE_ESTABLISHED || state==STATE_RECONNECTING)){
|
|
memmove(&rttHistory[1], rttHistory, 31*sizeof(double));
|
|
rttHistory[0]=GetAverageRTT();
|
|
/*if(rttHistory[16]>0){
|
|
LOGI("rtt diff: %.3lf", rttHistory[0]-rttHistory[16]);
|
|
}*/
|
|
int i;
|
|
double v=0;
|
|
for(i=1;i<32;i++){
|
|
v+=rttHistory[i-1]-rttHistory[i];
|
|
}
|
|
v=v/32;
|
|
if(rttHistory[0]>10.0 && rttHistory[8]>10.0 && (networkType==NET_TYPE_EDGE || networkType==NET_TYPE_GPRS)){
|
|
waitingForAcks=true;
|
|
signalBarCount=1;
|
|
}else{
|
|
waitingForAcks=false;
|
|
}
|
|
if(waitingForAcks)
|
|
wasWaitingForAcks=false;
|
|
//LOGI("%.3lf/%.3lf, rtt diff %.3lf, waiting=%d, queue=%d", rttHistory[0], rttHistory[8], v, waitingForAcks, sendQueue->Size());
|
|
if(jitterBuffer){
|
|
int lostCount=jitterBuffer->GetAndResetLostPacketCount();
|
|
if(lostCount>0 || (lostCount<0 && recvLossCount>((uint32_t)-lostCount)))
|
|
recvLossCount+=lostCount;
|
|
}
|
|
}
|
|
if(dontSendPackets>0)
|
|
dontSendPackets--;
|
|
|
|
int i;
|
|
|
|
conctl->Tick();
|
|
|
|
if(useTCP && !didAddTcpRelays){
|
|
std::vector<Endpoint *> relays;
|
|
for(std::vector<Endpoint *>::iterator itr=endpoints.begin(); itr!=endpoints.end(); ++itr){
|
|
if((*itr)->type!=EP_TYPE_UDP_RELAY)
|
|
continue;
|
|
Endpoint *tcpRelay=new Endpoint(**itr);
|
|
tcpRelay->type=EP_TYPE_TCP_RELAY;
|
|
tcpRelay->averageRTT=0;
|
|
tcpRelay->lastPingSeq=0;
|
|
tcpRelay->lastPingTime=0;
|
|
memset(tcpRelay->rtts, 0, sizeof(tcpRelay->rtts));
|
|
relays.push_back(tcpRelay);
|
|
}
|
|
endpoints.insert(endpoints.end(), relays.begin(), relays.end());
|
|
didAddTcpRelays=true;
|
|
}
|
|
|
|
if(state==STATE_ESTABLISHED && encoder && conctl){
|
|
if((audioInput && !audioInput->IsInitialized()) || (audioOutput && !audioOutput->IsInitialized())){
|
|
LOGE("Audio I/O failed");
|
|
lastError=TGVOIP_ERROR_AUDIO_IO;
|
|
SetState(STATE_FAILED);
|
|
}
|
|
|
|
int act=conctl->GetBandwidthControlAction();
|
|
if(act==TGVOIP_CONCTL_ACT_DECREASE){
|
|
uint32_t bitrate=encoder->GetBitrate();
|
|
if(bitrate>8000)
|
|
encoder->SetBitrate(bitrate<(minAudioBitrate+audioBitrateStepDecr) ? minAudioBitrate : (bitrate-audioBitrateStepDecr));
|
|
}else if(act==TGVOIP_CONCTL_ACT_INCREASE){
|
|
uint32_t bitrate=encoder->GetBitrate();
|
|
if(bitrate<maxBitrate)
|
|
encoder->SetBitrate(bitrate+audioBitrateStepIncr);
|
|
}
|
|
|
|
if(tickCount%10==0 && encoder){
|
|
uint32_t sendLossCount=conctl->GetSendLossCount();
|
|
memmove(sendLossCountHistory+1, sendLossCountHistory, 31*sizeof(uint32_t));
|
|
sendLossCountHistory[0]=sendLossCount-prevSendLossCount;
|
|
prevSendLossCount=sendLossCount;
|
|
double avgSendLossCount=0;
|
|
for(i=0;i<10;i++){
|
|
avgSendLossCount+=sendLossCountHistory[i];
|
|
}
|
|
double packetsPerSec=1000/(double)outgoingStreams[0]->frameDuration;
|
|
avgSendLossCount=avgSendLossCount/10/packetsPerSec;
|
|
//LOGV("avg send loss: %.1f%%", avgSendLossCount*100);
|
|
|
|
if(avgSendLossCount>0.1){
|
|
encoder->SetPacketLoss(40);
|
|
}else if(avgSendLossCount>0.075){
|
|
encoder->SetPacketLoss(35);
|
|
}else if(avgSendLossCount>0.0625){
|
|
encoder->SetPacketLoss(30);
|
|
}else if(avgSendLossCount>0.05){
|
|
encoder->SetPacketLoss(25);
|
|
}else if(avgSendLossCount>0.025){
|
|
encoder->SetPacketLoss(20);
|
|
}else if(avgSendLossCount>0.01){
|
|
encoder->SetPacketLoss(17);
|
|
}else{
|
|
encoder->SetPacketLoss(15);
|
|
}
|
|
|
|
if(encoder->GetPacketLoss()>30)
|
|
signalBarCount=MIN(signalBarCount, 2);
|
|
else if(encoder->GetPacketLoss()>20)
|
|
signalBarCount=MIN(signalBarCount, 3);
|
|
}
|
|
}
|
|
|
|
bool areThereAnyEnabledStreams=false;
|
|
|
|
for(i=0;i<outgoingStreams.size();i++){
|
|
if(outgoingStreams[i]->enabled)
|
|
areThereAnyEnabledStreams=true;
|
|
}
|
|
|
|
if((waitingForAcks && tickCount%10==0) || (!areThereAnyEnabledStreams && tickCount%2==0)){
|
|
unsigned char* buf=outgoingPacketsBufferPool.Get();
|
|
if(buf){
|
|
sendQueue->Put(PendingOutgoingPacket{
|
|
/*.seq=*/(firstSentPing=GenerateOutSeq()),
|
|
/*.type=*/PKT_NOP,
|
|
/*.len=*/0,
|
|
/*.data=*/buf,
|
|
/*.endpoint=*/NULL
|
|
});
|
|
}
|
|
}
|
|
|
|
if(state==STATE_WAIT_INIT_ACK && GetCurrentTime()-stateChangeTime>.5){
|
|
SendInit();
|
|
}
|
|
|
|
if(waitingForRelayPeerInfo && GetCurrentTime()-publicEndpointsReqTime>5){
|
|
LOGD("Resending peer relay info request");
|
|
SendPublicEndpointsRequest();
|
|
}
|
|
|
|
lock_mutex(queuedPacketsMutex);
|
|
for(i=0;i<queuedPackets.size();i++){
|
|
voip_queued_packet_t* qp=queuedPackets[i];
|
|
if(qp->timeout>0 && qp->firstSentTime>0 && GetCurrentTime()-qp->firstSentTime>=qp->timeout){
|
|
LOGD("Removing queued packet because of timeout");
|
|
if(qp->data)
|
|
free(qp->data);
|
|
free(qp);
|
|
queuedPackets.erase(queuedPackets.begin()+i);
|
|
i--;
|
|
continue;
|
|
}
|
|
if(GetCurrentTime()-qp->lastSentTime>=qp->retryInterval){
|
|
unsigned char* buf=outgoingPacketsBufferPool.Get();
|
|
if(buf){
|
|
uint32_t seq=GenerateOutSeq();
|
|
memmove(&qp->seqs[1], qp->seqs, 4*9);
|
|
qp->seqs[0]=seq;
|
|
qp->lastSentTime=GetCurrentTime();
|
|
LOGD("Sending queued packet, seq=%u, type=%u, len=%u", seq, qp->type, unsigned(qp->length));
|
|
if(qp->firstSentTime==0)
|
|
qp->firstSentTime=qp->lastSentTime;
|
|
if(qp->length)
|
|
memcpy(buf, qp->data, qp->length);
|
|
sendQueue->Put(PendingOutgoingPacket{
|
|
/*.seq=*/seq,
|
|
/*.type=*/qp->type,
|
|
/*.len=*/qp->length,
|
|
/*.data=*/buf,
|
|
/*.endpoint=*/NULL
|
|
});
|
|
}
|
|
}
|
|
}
|
|
unlock_mutex(queuedPacketsMutex);
|
|
|
|
if(jitterBuffer){
|
|
jitterBuffer->Tick();
|
|
double avgDelay=jitterBuffer->GetAverageDelay();
|
|
double avgLateCount[3];
|
|
jitterBuffer->GetAverageLateCount(avgLateCount);
|
|
/*if(avgDelay>=5)
|
|
signalBarCount=1;
|
|
else if(avgDelay>=4)
|
|
signalBarCount=MIN(signalBarCount, 2);
|
|
else if(avgDelay>=3)
|
|
signalBarCount=MIN(signalBarCount, 3);*/
|
|
|
|
if(avgLateCount[2]>=0.2)
|
|
signalBarCount=1;
|
|
else if(avgLateCount[2]>=0.1)
|
|
signalBarCount=MIN(signalBarCount, 2);
|
|
|
|
}
|
|
|
|
lock_mutex(endpointsMutex);
|
|
if(state==STATE_ESTABLISHED || state==STATE_RECONNECTING){
|
|
Endpoint* minPingRelay=preferredRelay;
|
|
double minPing=preferredRelay->averageRTT;
|
|
for(std::vector<Endpoint*>::iterator e=endpoints.begin();e!=endpoints.end();++e){
|
|
Endpoint* endpoint=*e;
|
|
if(endpoint->type==EP_TYPE_TCP_RELAY && !useTCP)
|
|
continue;
|
|
if(GetCurrentTime()-endpoint->lastPingTime>=10){
|
|
LOGV("Sending ping to %s", endpoint->address.ToString().c_str());
|
|
unsigned char* buf=outgoingPacketsBufferPool.Get();
|
|
if(buf){
|
|
sendQueue->Put(PendingOutgoingPacket{
|
|
/*.seq=*/(endpoint->lastPingSeq=GenerateOutSeq()),
|
|
/*.type=*/PKT_PING,
|
|
/*.len=*/0,
|
|
/*.data=*/buf,
|
|
/*.endpoint=*/endpoint
|
|
});
|
|
}
|
|
endpoint->lastPingTime=GetCurrentTime();
|
|
}
|
|
if(endpoint->type==EP_TYPE_UDP_RELAY || (useTCP && endpoint->type==EP_TYPE_TCP_RELAY)){
|
|
double k=endpoint->type==EP_TYPE_UDP_RELAY ? 1 : 2;
|
|
if(endpoint->averageRTT>0 && endpoint->averageRTT*k<minPing*relaySwitchThreshold){
|
|
minPing=endpoint->averageRTT*k;
|
|
minPingRelay=endpoint;
|
|
}
|
|
}
|
|
}
|
|
if(minPingRelay!=preferredRelay){
|
|
preferredRelay=minPingRelay;
|
|
LOGV("set preferred relay to %s", preferredRelay->address.ToString().c_str());
|
|
if(currentEndpoint->type==EP_TYPE_UDP_RELAY || currentEndpoint->type==EP_TYPE_TCP_RELAY)
|
|
currentEndpoint=preferredRelay;
|
|
LogDebugInfo();
|
|
/*BufferOutputStream pkt(32);
|
|
pkt.WriteInt64(preferredRelay->id);
|
|
SendPacketReliably(PKT_SWITCH_PREF_RELAY, pkt.GetBuffer(), pkt.GetLength(), 1, 9);*/
|
|
}
|
|
if(currentEndpoint->type==EP_TYPE_UDP_RELAY){
|
|
Endpoint* p2p=GetEndpointByType(EP_TYPE_UDP_P2P_INET);
|
|
if(p2p){
|
|
Endpoint* lan=GetEndpointByType(EP_TYPE_UDP_P2P_LAN);
|
|
if(lan && lan->averageRTT>0 && lan->averageRTT<minPing*relayToP2pSwitchThreshold){
|
|
//SendPacketReliably(PKT_SWITCH_TO_P2P, NULL, 0, 1, 5);
|
|
currentEndpoint=lan;
|
|
LOGI("Switching to p2p (LAN)");
|
|
LogDebugInfo();
|
|
}else{
|
|
if(p2p->averageRTT>0 && p2p->averageRTT<minPing*relayToP2pSwitchThreshold){
|
|
//SendPacketReliably(PKT_SWITCH_TO_P2P, NULL, 0, 1, 5);
|
|
currentEndpoint=p2p;
|
|
LOGI("Switching to p2p (Inet)");
|
|
LogDebugInfo();
|
|
}
|
|
}
|
|
}
|
|
}else{
|
|
if(minPing>0 && minPing<currentEndpoint->averageRTT*p2pToRelaySwitchThreshold){
|
|
LOGI("Switching to relay");
|
|
currentEndpoint=preferredRelay;
|
|
LogDebugInfo();
|
|
}
|
|
}
|
|
}
|
|
if(udpConnectivityState==UDP_UNKNOWN){
|
|
for(std::vector<Endpoint*>::iterator itr=endpoints.begin();itr!=endpoints.end();++itr){
|
|
if((*itr)->type==EP_TYPE_UDP_RELAY){
|
|
SendUdpPing(*itr);
|
|
}
|
|
}
|
|
udpConnectivityState=UDP_PING_SENT;
|
|
lastUdpPingTime=time;
|
|
udpPingCount=1;
|
|
}else if(udpConnectivityState==UDP_PING_SENT){
|
|
if(time-lastUdpPingTime>=0.5){
|
|
if(udpPingCount<4){
|
|
for(std::vector<Endpoint*>::iterator itr=endpoints.begin();itr!=endpoints.end();++itr){
|
|
if((*itr)->type==EP_TYPE_UDP_RELAY){
|
|
SendUdpPing(*itr);
|
|
}
|
|
}
|
|
udpPingCount++;
|
|
lastUdpPingTime=time;
|
|
}else{
|
|
LOGW("No UDP ping replies received; assuming no connectivity and trying TCP")
|
|
udpConnectivityState=UDP_NOT_AVAILABLE;
|
|
useTCP=true;
|
|
}
|
|
}
|
|
}
|
|
unlock_mutex(endpointsMutex);
|
|
|
|
if(state==STATE_ESTABLISHED || state==STATE_RECONNECTING){
|
|
if(time-lastRecvPacketTime>=config.recv_timeout){
|
|
if(currentEndpoint && currentEndpoint->type!=EP_TYPE_UDP_RELAY && currentEndpoint->type!=EP_TYPE_TCP_RELAY){
|
|
LOGW("Packet receive timeout, switching to relay");
|
|
currentEndpoint=preferredRelay;
|
|
for(std::vector<Endpoint*>::iterator itrtr=endpoints.begin();itrtr!=endpoints.end();++itrtr){
|
|
Endpoint* e=*itrtr;
|
|
if(e->type==EP_TYPE_UDP_P2P_INET || e->type==EP_TYPE_UDP_P2P_LAN){
|
|
e->averageRTT=0;
|
|
memset(e->rtts, 0, sizeof(e->rtts));
|
|
}
|
|
}
|
|
if(allowP2p){
|
|
SendPublicEndpointsRequest();
|
|
}
|
|
UpdateDataSavingState();
|
|
UpdateAudioBitrate();
|
|
BufferOutputStream s(4);
|
|
s.WriteInt32(dataSavingMode ? INIT_FLAG_DATA_SAVING_ENABLED : 0);
|
|
SendPacketReliably(PKT_NETWORK_CHANGED, s.GetBuffer(), s.GetLength(), 1, 20);
|
|
lastRecvPacketTime=time;
|
|
}else{
|
|
LOGW("Packet receive timeout, disconnecting");
|
|
lastError=TGVOIP_ERROR_TIMEOUT;
|
|
SetState(STATE_FAILED);
|
|
}
|
|
}
|
|
}else if(state==STATE_WAIT_INIT || state==STATE_WAIT_INIT_ACK){
|
|
if(GetCurrentTime()-connectionInitTime>=config.init_timeout){
|
|
LOGW("Init timeout, disconnecting");
|
|
lastError=TGVOIP_ERROR_TIMEOUT;
|
|
SetState(STATE_FAILED);
|
|
}
|
|
}
|
|
|
|
if(state==STATE_ESTABLISHED && time-lastRecvPacketTime>=reconnectingTimeout){
|
|
SetState(STATE_RECONNECTING);
|
|
}
|
|
|
|
if(state!=STATE_ESTABLISHED && setEstablishedAt>0 && time>=setEstablishedAt){
|
|
SetState(STATE_ESTABLISHED);
|
|
setEstablishedAt=0;
|
|
}
|
|
|
|
signalBarsHistory[tickCount%sizeof(signalBarsHistory)]=(unsigned char)signalBarCount;
|
|
int _signalBarCount=GetSignalBarsCount();
|
|
if(_signalBarCount!=prevSignalBarCount){
|
|
LOGD("SIGNAL BAR COUNT CHANGED: %d", _signalBarCount);
|
|
if(signalBarCountCallback)
|
|
signalBarCountCallback(this, _signalBarCount);
|
|
}
|
|
|
|
|
|
if(statsDump){
|
|
//fprintf(statsDump, "Time\tRTT\tLISeq\tLASeq\tCWnd\tBitrate\tJitter\tJDelay\tAJDelay\n");
|
|
fprintf(statsDump, "%.3f\t%.3f\t%d\t%d\t%d\t%d\t%d\t%d\t%d\t%d\t%.3f\t%.3f\t%.3f\n",
|
|
GetCurrentTime()-startTime,
|
|
currentEndpoint->rtts[0],
|
|
lastRemoteSeq,
|
|
seq,
|
|
lastRemoteAckSeq,
|
|
recvLossCount,
|
|
conctl ? conctl->GetSendLossCount() : 0,
|
|
conctl ? (int)conctl->GetInflightDataSize() : 0,
|
|
encoder ? encoder->GetBitrate() : 0,
|
|
encoder ? encoder->GetPacketLoss() : 0,
|
|
jitterBuffer ? jitterBuffer->GetLastMeasuredJitter() : 0,
|
|
jitterBuffer ? jitterBuffer->GetLastMeasuredDelay()*0.06 : 0,
|
|
jitterBuffer ? jitterBuffer->GetAverageDelay()*0.06 : 0);
|
|
}
|
|
|
|
#if defined(__APPLE__) && defined(TGVOIP_USE_AUDIO_SESSION)
|
|
if(needNotifyAcquiredAudioSession){
|
|
needNotifyAcquiredAudioSession=false;
|
|
audio::AudioUnitIO::AudioSessionAcquired();
|
|
}
|
|
#endif
|
|
}
|
|
LOGI("=== tick thread exiting ===");
|
|
}
|
|
|
|
|
|
Endpoint& VoIPController::GetRemoteEndpoint(){
|
|
//return useLan ? &remoteLanEp : &remotePublicEp;
|
|
return *currentEndpoint;
|
|
}
|
|
|
|
|
|
void VoIPController::SendPacket(unsigned char *data, size_t len, Endpoint* ep){
|
|
if(stopping)
|
|
return;
|
|
if(ep->type==EP_TYPE_TCP_RELAY && !useTCP)
|
|
return;
|
|
//dst.sin_addr=ep->address;
|
|
//dst.sin_port=htons(ep->port);
|
|
//dst.sin_family=AF_INET;
|
|
BufferOutputStream out(len+128);
|
|
if(ep->type==EP_TYPE_UDP_RELAY || ep->type==EP_TYPE_TCP_RELAY)
|
|
out.WriteBytes((unsigned char*)ep->peerTag, 16);
|
|
else
|
|
out.WriteBytes(callID, 16);
|
|
if(len>0){
|
|
BufferOutputStream inner(len+128);
|
|
inner.WriteInt32(len);
|
|
inner.WriteBytes(data, len);
|
|
if(inner.GetLength()%16!=0){
|
|
size_t padLen=16-inner.GetLength()%16;
|
|
unsigned char padding[16];
|
|
crypto.rand_bytes((uint8_t *) padding, padLen);
|
|
inner.WriteBytes(padding, padLen);
|
|
}
|
|
assert(inner.GetLength()%16==0);
|
|
unsigned char key[32], iv[32], msgHash[SHA1_LENGTH];
|
|
crypto.sha1((uint8_t *) inner.GetBuffer(), len+4, msgHash);
|
|
out.WriteBytes(keyFingerprint, 8);
|
|
out.WriteBytes((msgHash+(SHA1_LENGTH-16)), 16);
|
|
KDF(msgHash+(SHA1_LENGTH-16), isOutgoing ? 0 : 8, key, iv);
|
|
unsigned char aesOut[MSC_STACK_FALLBACK(inner.GetLength(), 1500)];
|
|
crypto.aes_ige_encrypt(inner.GetBuffer(), aesOut, inner.GetLength(), key, iv);
|
|
out.WriteBytes(aesOut, inner.GetLength());
|
|
}
|
|
//LOGV("Sending %d bytes to %s:%d", out.GetLength(), ep->address.ToString().c_str(), ep->port);
|
|
if(IS_MOBILE_NETWORK(networkType))
|
|
stats.bytesSentMobile+=(uint64_t)out.GetLength();
|
|
else
|
|
stats.bytesSentWifi+=(uint64_t)out.GetLength();
|
|
|
|
NetworkPacket pkt;
|
|
//LOGE("WILL SEND PACKET OF SIZE %i", out.GetLength());
|
|
pkt.address=(NetworkAddress*)&ep->address;
|
|
pkt.port=ep->port;
|
|
pkt.length=out.GetLength();
|
|
pkt.data=out.GetBuffer();
|
|
pkt.protocol=ep->type==EP_TYPE_TCP_RELAY ? PROTO_TCP : PROTO_UDP;
|
|
//socket->Send(&pkt);
|
|
if(ep->type==EP_TYPE_TCP_RELAY){
|
|
if(ep->socket){
|
|
ep->socket->Send(&pkt);
|
|
}else{
|
|
LOGI("connecting to tcp: %s:%u", ep->address.ToString().c_str(), ep->port);
|
|
NetworkSocket* s;
|
|
if(proxyProtocol==PROXY_NONE){
|
|
s=NetworkSocket::Create(PROTO_TCP);
|
|
}else if(proxyProtocol==PROXY_SOCKS5){
|
|
NetworkSocket* rawTcp=NetworkSocket::Create(PROTO_TCP);
|
|
openingTcpSocket=rawTcp;
|
|
rawTcp->Connect(resolvedProxyAddress, proxyPort);
|
|
if(rawTcp->IsFailed()){
|
|
openingTcpSocket=NULL;
|
|
rawTcp->Close();
|
|
delete rawTcp;
|
|
LOGW("Error connecting to SOCKS5 proxy");
|
|
return;
|
|
}
|
|
NetworkSocketSOCKS5Proxy* proxy=new NetworkSocketSOCKS5Proxy(rawTcp, NULL, proxyUsername, proxyPassword);
|
|
openingTcpSocket=rawTcp;
|
|
proxy->InitConnection();
|
|
if(proxy->IsFailed()){
|
|
openingTcpSocket=NULL;
|
|
LOGW("Proxy initialization failed");
|
|
proxy->Close();
|
|
delete proxy;
|
|
return;
|
|
}
|
|
s=proxy;
|
|
}/*else if(proxyProtocol==PROXY_HTTP){
|
|
s=NetworkSocket::Create(PROTO_TCP);
|
|
}*/else{
|
|
LOGE("Unsupported proxy protocol %d", proxyProtocol);
|
|
SetState(STATE_FAILED);
|
|
return;
|
|
}
|
|
s->Connect(&ep->address, ep->port);
|
|
if(s->IsFailed()){
|
|
s->Close();
|
|
delete s;
|
|
LOGW("Error connecting to %s:%u", ep->address.ToString().c_str(), ep->port);
|
|
}else{
|
|
NetworkSocketTCPObfuscated* tcpWrapper=new NetworkSocketTCPObfuscated(s);
|
|
openingTcpSocket=tcpWrapper;
|
|
tcpWrapper->InitConnection();
|
|
openingTcpSocket=NULL;
|
|
if(tcpWrapper->IsFailed()){
|
|
tcpWrapper->Close();
|
|
delete tcpWrapper;
|
|
LOGW("Error initializing connection to %s:%u", ep->address.ToString().c_str(), ep->port);
|
|
}else{
|
|
tcpWrapper->Send(&pkt);
|
|
ep->socket=tcpWrapper;
|
|
selectCanceller->CancelSelect();
|
|
}
|
|
}
|
|
}
|
|
}else{
|
|
udpSocket->Send(&pkt);
|
|
}
|
|
}
|
|
|
|
|
|
void VoIPController::SetNetworkType(int type){
|
|
networkType=type;
|
|
UpdateDataSavingState();
|
|
UpdateAudioBitrate();
|
|
std::string itfName=udpSocket->GetLocalInterfaceInfo(NULL, NULL);
|
|
if(itfName!=activeNetItfName){
|
|
udpSocket->OnActiveInterfaceChanged();
|
|
LOGI("Active network interface changed: %s -> %s", activeNetItfName.c_str(), itfName.c_str());
|
|
bool isFirstChange=activeNetItfName.length()==0;
|
|
activeNetItfName=itfName;
|
|
if(isFirstChange)
|
|
return;
|
|
if(currentEndpoint && currentEndpoint->type!=EP_TYPE_UDP_RELAY){
|
|
if(preferredRelay->type==EP_TYPE_UDP_RELAY)
|
|
currentEndpoint=preferredRelay;
|
|
lock_mutex(endpointsMutex);
|
|
for(std::vector<Endpoint*>::iterator itr=endpoints.begin();itr!=endpoints.end();){
|
|
Endpoint* endpoint=*itr;
|
|
if(endpoint->type==EP_TYPE_UDP_RELAY && useTCP){
|
|
useTCP=false;
|
|
if(preferredRelay->type==EP_TYPE_TCP_RELAY){
|
|
preferredRelay=endpoint;
|
|
currentEndpoint=endpoint;
|
|
}
|
|
}else if(endpoint->type==EP_TYPE_TCP_RELAY && endpoint->socket){
|
|
endpoint->socket->Close();
|
|
}
|
|
//if(endpoint->type==EP_TYPE_UDP_P2P_INET){
|
|
endpoint->averageRTT=0;
|
|
memset(endpoint->rtts, 0, sizeof(endpoint->rtts));
|
|
//}
|
|
if(endpoint->type==EP_TYPE_UDP_P2P_LAN){
|
|
delete endpoint;
|
|
itr=endpoints.erase(itr);
|
|
}else{
|
|
++itr;
|
|
}
|
|
}
|
|
unlock_mutex(endpointsMutex);
|
|
}
|
|
udpConnectivityState=UDP_UNKNOWN;
|
|
udpPingCount=0;
|
|
lastUdpPingTime=0;
|
|
if(proxyProtocol==PROXY_SOCKS5)
|
|
InitUDPProxy();
|
|
if(allowP2p && currentEndpoint){
|
|
SendPublicEndpointsRequest();
|
|
}
|
|
BufferOutputStream s(4);
|
|
s.WriteInt32(dataSavingMode ? INIT_FLAG_DATA_SAVING_ENABLED : 0);
|
|
SendPacketReliably(PKT_NETWORK_CHANGED, s.GetBuffer(), s.GetLength(), 1, 20);
|
|
selectCanceller->CancelSelect();
|
|
}
|
|
LOGI("set network type: %d, active interface %s", type, activeNetItfName.c_str());
|
|
/*if(type==NET_TYPE_GPRS || type==NET_TYPE_EDGE)
|
|
audioPacketGrouping=2;
|
|
else
|
|
audioPacketGrouping=1;*/
|
|
}
|
|
|
|
|
|
double VoIPController::GetAverageRTT(){
|
|
if(lastSentSeq>=lastRemoteAckSeq){
|
|
uint32_t diff=lastSentSeq-lastRemoteAckSeq;
|
|
//LOGV("rtt diff=%u", diff);
|
|
if(diff<32){
|
|
int i;
|
|
double res=0;
|
|
int count=0;
|
|
for(i=diff;i<32;i++){
|
|
if(remoteAcks[i-diff]>0){
|
|
res+=(remoteAcks[i-diff]-sentPacketTimes[i]);
|
|
count++;
|
|
}
|
|
}
|
|
if(count>0)
|
|
res/=count;
|
|
return res;
|
|
}
|
|
}
|
|
return 999;
|
|
}
|
|
|
|
#if defined(__APPLE__)
|
|
static void initMachTimestart() {
|
|
mach_timebase_info_data_t tb = { 0, 0 };
|
|
mach_timebase_info(&tb);
|
|
VoIPController::machTimebase = tb.numer;
|
|
VoIPController::machTimebase /= tb.denom;
|
|
VoIPController::machTimestart = mach_absolute_time();
|
|
}
|
|
#endif
|
|
|
|
double VoIPController::GetCurrentTime(){
|
|
#if defined(__linux__)
|
|
struct timespec ts;
|
|
clock_gettime(CLOCK_MONOTONIC, &ts);
|
|
return ts.tv_sec+(double)ts.tv_nsec/1000000000.0;
|
|
#elif defined(__APPLE__)
|
|
static pthread_once_t token = PTHREAD_ONCE_INIT;
|
|
pthread_once(&token, &initMachTimestart);
|
|
return (mach_absolute_time() - machTimestart) * machTimebase / 1000000000.0f;
|
|
#elif defined(_WIN32)
|
|
if(!didInitWin32TimeScale){
|
|
LARGE_INTEGER scale;
|
|
QueryPerformanceFrequency(&scale);
|
|
win32TimeScale=scale.QuadPart;
|
|
didInitWin32TimeScale=true;
|
|
}
|
|
LARGE_INTEGER t;
|
|
QueryPerformanceCounter(&t);
|
|
return (double)t.QuadPart/(double)win32TimeScale;
|
|
#endif
|
|
}
|
|
|
|
void VoIPController::SetStateCallback(void (* f)(VoIPController*, int)){
|
|
stateCallback=f;
|
|
if(stateCallback){
|
|
stateCallback(this, state);
|
|
}
|
|
}
|
|
|
|
|
|
void VoIPController::SetState(int state){
|
|
this->state=state;
|
|
LOGV("Call state changed to %d", state);
|
|
stateChangeTime=GetCurrentTime();
|
|
if(stateCallback){
|
|
stateCallback(this, state);
|
|
}
|
|
}
|
|
|
|
|
|
void VoIPController::SetMicMute(bool mute){
|
|
micMuted=mute;
|
|
if(audioInput){
|
|
if(mute)
|
|
audioInput->Stop();
|
|
else
|
|
audioInput->Start();
|
|
if(!audioInput->IsInitialized()){
|
|
lastError=TGVOIP_ERROR_AUDIO_IO;
|
|
SetState(STATE_FAILED);
|
|
return;
|
|
}
|
|
}
|
|
if(echoCanceller)
|
|
echoCanceller->Enable(!mute);
|
|
int i;
|
|
for(i=0;i<outgoingStreams.size();i++){
|
|
if(outgoingStreams[i]->type==STREAM_TYPE_AUDIO){
|
|
unsigned char buf[2];
|
|
buf[0]=outgoingStreams[i]->id;
|
|
buf[1]=(char) (mute ? 0 : 1);
|
|
SendPacketReliably(PKT_STREAM_STATE, buf, 2, .5f, 20);
|
|
outgoingStreams[i]->enabled=!mute;
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
void VoIPController::UpdateAudioOutputState(){
|
|
bool areAnyAudioStreamsEnabled=false;
|
|
int i;
|
|
for(i=0;i<incomingStreams.size();i++){
|
|
if(incomingStreams[i]->type==STREAM_TYPE_AUDIO && incomingStreams[i]->enabled)
|
|
areAnyAudioStreamsEnabled=true;
|
|
}
|
|
if(jitterBuffer){
|
|
jitterBuffer->Reset();
|
|
}
|
|
if(decoder){
|
|
decoder->ResetQueue();
|
|
}
|
|
if(audioOutput){
|
|
if(audioOutput->IsPlaying()!=areAnyAudioStreamsEnabled){
|
|
if(areAnyAudioStreamsEnabled)
|
|
audioOutput->Start();
|
|
else
|
|
audioOutput->Stop();
|
|
}
|
|
}
|
|
}
|
|
|
|
void VoIPController::KDF(unsigned char* msgKey, size_t x, unsigned char* aesKey, unsigned char* aesIv){
|
|
uint8_t sA[SHA1_LENGTH], sB[SHA1_LENGTH], sC[SHA1_LENGTH], sD[SHA1_LENGTH];
|
|
BufferOutputStream buf(128);
|
|
buf.WriteBytes(msgKey, 16);
|
|
buf.WriteBytes(encryptionKey+x, 32);
|
|
crypto.sha1(buf.GetBuffer(), buf.GetLength(), sA);
|
|
buf.Reset();
|
|
buf.WriteBytes(encryptionKey+32+x, 16);
|
|
buf.WriteBytes(msgKey, 16);
|
|
buf.WriteBytes(encryptionKey+48+x, 16);
|
|
crypto.sha1(buf.GetBuffer(), buf.GetLength(), sB);
|
|
buf.Reset();
|
|
buf.WriteBytes(encryptionKey+64+x, 32);
|
|
buf.WriteBytes(msgKey, 16);
|
|
crypto.sha1(buf.GetBuffer(), buf.GetLength(), sC);
|
|
buf.Reset();
|
|
buf.WriteBytes(msgKey, 16);
|
|
buf.WriteBytes(encryptionKey+96+x, 32);
|
|
crypto.sha1(buf.GetBuffer(), buf.GetLength(), sD);
|
|
buf.Reset();
|
|
buf.WriteBytes(sA, 8);
|
|
buf.WriteBytes(sB+8, 12);
|
|
buf.WriteBytes(sC+4, 12);
|
|
assert(buf.GetLength()==32);
|
|
memcpy(aesKey, buf.GetBuffer(), 32);
|
|
buf.Reset();
|
|
buf.WriteBytes(sA+8, 12);
|
|
buf.WriteBytes(sB, 8);
|
|
buf.WriteBytes(sC+16, 4);
|
|
buf.WriteBytes(sD, 8);
|
|
assert(buf.GetLength()==32);
|
|
memcpy(aesIv, buf.GetBuffer(), 32);
|
|
}
|
|
|
|
void VoIPController::GetDebugString(char *buffer, size_t len){
|
|
char endpointsBuf[10240];
|
|
memset(endpointsBuf, 0, 10240);
|
|
int i;
|
|
for(std::vector<Endpoint*>::iterator itrtr=endpoints.begin();itrtr!=endpoints.end();++itrtr){
|
|
const char* type;
|
|
Endpoint* endpoint=*itrtr;
|
|
switch(endpoint->type){
|
|
case EP_TYPE_UDP_P2P_INET:
|
|
type="UDP_P2P_INET";
|
|
break;
|
|
case EP_TYPE_UDP_P2P_LAN:
|
|
type="UDP_P2P_LAN";
|
|
break;
|
|
case EP_TYPE_UDP_RELAY:
|
|
type="UDP_RELAY";
|
|
break;
|
|
case EP_TYPE_TCP_RELAY:
|
|
type="TCP_RELAY";
|
|
break;
|
|
default:
|
|
type="UNKNOWN";
|
|
break;
|
|
}
|
|
if(strlen(endpointsBuf)>10240-1024)
|
|
break;
|
|
sprintf(endpointsBuf+strlen(endpointsBuf), "%s:%u %dms [%s%s]\n", endpoint->address.ToString().c_str(), endpoint->port, (int)(endpoint->averageRTT*1000), type, currentEndpoint==endpoint ? ", IN_USE" : "");
|
|
}
|
|
double avgLate[3];
|
|
if(jitterBuffer)
|
|
jitterBuffer->GetAverageLateCount(avgLate);
|
|
else
|
|
memset(avgLate, 0, 3*sizeof(double));
|
|
snprintf(buffer, len,
|
|
"Remote endpoints: \n%s"
|
|
"Jitter buffer: %d/%.2f | %.1f, %.1f, %.1f\n"
|
|
"RTT avg/min: %d/%d\n"
|
|
"Congestion window: %d/%d bytes\n"
|
|
"Key fingerprint: %02hhX%02hhX%02hhX%02hhX%02hhX%02hhX%02hhX%02hhX\n"
|
|
"Last sent/ack'd seq: %u/%u\n"
|
|
"Last recvd seq: %u\n"
|
|
"Send/recv losses: %u/%u (%d%%)\n"
|
|
"Audio bitrate: %d kbit\n"
|
|
// "Packet grouping: %d\n"
|
|
"Frame size out/in: %d/%d\n"
|
|
"Bytes sent/recvd: %llu/%llu",
|
|
endpointsBuf,
|
|
jitterBuffer ? jitterBuffer->GetMinPacketCount() : 0, jitterBuffer ? jitterBuffer->GetAverageDelay() : 0, avgLate[0], avgLate[1], avgLate[2],
|
|
// (int)(GetAverageRTT()*1000), 0,
|
|
(int)(conctl->GetAverageRTT()*1000), (int)(conctl->GetMinimumRTT()*1000),
|
|
int(conctl->GetInflightDataSize()), int(conctl->GetCongestionWindow()),
|
|
keyFingerprint[0],keyFingerprint[1],keyFingerprint[2],keyFingerprint[3],keyFingerprint[4],keyFingerprint[5],keyFingerprint[6],keyFingerprint[7],
|
|
lastSentSeq, lastRemoteAckSeq, lastRemoteSeq,
|
|
conctl->GetSendLossCount(), recvLossCount, encoder ? encoder->GetPacketLoss() : 0,
|
|
encoder ? (encoder->GetBitrate()/1000) : 0,
|
|
// audioPacketGrouping,
|
|
outgoingStreams[0]->frameDuration, incomingStreams.size()>0 ? incomingStreams[0]->frameDuration : 0,
|
|
(long long unsigned int)(stats.bytesSentMobile+stats.bytesSentWifi),
|
|
(long long unsigned int)(stats.bytesRecvdMobile+stats.bytesRecvdWifi));
|
|
}
|
|
|
|
|
|
void VoIPController::SendPublicEndpointsRequest(){
|
|
LOGI("Sending public endpoints request");
|
|
if(preferredRelay){
|
|
SendPublicEndpointsRequest(*preferredRelay);
|
|
}
|
|
if(peerPreferredRelay && peerPreferredRelay!=preferredRelay){
|
|
SendPublicEndpointsRequest(*peerPreferredRelay);
|
|
}
|
|
}
|
|
|
|
void VoIPController::SendPublicEndpointsRequest(Endpoint& relay){
|
|
if(!useUDP)
|
|
return;
|
|
LOGD("Sending public endpoints request to %s:%d", relay.address.ToString().c_str(), relay.port);
|
|
publicEndpointsReqTime=GetCurrentTime();
|
|
waitingForRelayPeerInfo=true;
|
|
unsigned char buf[32];
|
|
memcpy(buf, relay.peerTag, 16);
|
|
memset(buf+16, 0xFF, 16);
|
|
NetworkPacket pkt;
|
|
pkt.data=buf;
|
|
pkt.length=32;
|
|
pkt.address=(NetworkAddress*)&relay.address;
|
|
pkt.port=relay.port;
|
|
pkt.protocol=PROTO_UDP;
|
|
udpSocket->Send(&pkt);
|
|
}
|
|
|
|
Endpoint* VoIPController::GetEndpointByType(int type){
|
|
if(type==EP_TYPE_UDP_RELAY && preferredRelay)
|
|
return preferredRelay;
|
|
for(std::vector<Endpoint*>::iterator itrtr=endpoints.begin();itrtr!=endpoints.end();++itrtr){
|
|
if((*itrtr)->type==type)
|
|
return *itrtr;
|
|
}
|
|
return NULL;
|
|
}
|
|
|
|
|
|
float VoIPController::GetOutputLevel(){
|
|
if(!audioOutput || !audioOutStarted){
|
|
return 0.0;
|
|
}
|
|
return audioOutput->GetLevel();
|
|
}
|
|
|
|
|
|
void VoIPController::SendPacketReliably(unsigned char type, unsigned char *data, size_t len, double retryInterval, double timeout){
|
|
LOGD("Send reliably, type=%u, len=%u, retry=%.3f, timeout=%.3f", type, unsigned(len), retryInterval, timeout);
|
|
voip_queued_packet_t* pkt=(voip_queued_packet_t *) malloc(sizeof(voip_queued_packet_t));
|
|
memset(pkt, 0, sizeof(voip_queued_packet_t));
|
|
pkt->type=type;
|
|
if(data){
|
|
pkt->data=(unsigned char *) malloc(len);
|
|
memcpy(pkt->data, data, len);
|
|
pkt->length=len;
|
|
}
|
|
pkt->retryInterval=retryInterval;
|
|
pkt->timeout=timeout;
|
|
pkt->firstSentTime=0;
|
|
pkt->lastSentTime=0;
|
|
lock_mutex(queuedPacketsMutex);
|
|
queuedPackets.push_back(pkt);
|
|
unlock_mutex(queuedPacketsMutex);
|
|
}
|
|
|
|
|
|
void VoIPController::SetConfig(voip_config_t *cfg){
|
|
memcpy(&config, cfg, sizeof(voip_config_t));
|
|
if(tgvoipLogFile){
|
|
fclose(tgvoipLogFile);
|
|
}
|
|
if(strlen(cfg->logFilePath)){
|
|
tgvoipLogFile=fopen(cfg->logFilePath, "a");
|
|
tgvoip_log_file_write_header();
|
|
}
|
|
if(statsDump)
|
|
fclose(statsDump);
|
|
if(strlen(cfg->statsDumpFilePath)){
|
|
statsDump=fopen(cfg->statsDumpFilePath, "w");
|
|
if(statsDump)
|
|
fprintf(statsDump, "Time\tRTT\tLRSeq\tLSSeq\tLASeq\tLostR\tLostS\tCWnd\tBitrate\tLoss%%\tJitter\tJDelay\tAJDelay\n");
|
|
else
|
|
LOGW("Failed to open stats dump file %s for writing", cfg->statsDumpFilePath);
|
|
}
|
|
UpdateDataSavingState();
|
|
UpdateAudioBitrate();
|
|
}
|
|
|
|
|
|
void VoIPController::UpdateDataSavingState(){
|
|
if(config.data_saving==DATA_SAVING_ALWAYS){
|
|
dataSavingMode=true;
|
|
}else if(config.data_saving==DATA_SAVING_MOBILE){
|
|
dataSavingMode=networkType==NET_TYPE_GPRS || networkType==NET_TYPE_EDGE ||
|
|
networkType==NET_TYPE_3G || networkType==NET_TYPE_HSPA || networkType==NET_TYPE_LTE || networkType==NET_TYPE_OTHER_MOBILE;
|
|
}else{
|
|
dataSavingMode=false;
|
|
}
|
|
LOGI("update data saving mode, config %d, enabled %d, reqd by peer %d", config.data_saving, dataSavingMode, dataSavingRequestedByPeer);
|
|
}
|
|
|
|
|
|
void VoIPController::DebugCtl(int request, int param){
|
|
if(request==1){ // set bitrate
|
|
maxBitrate=param;
|
|
if(encoder){
|
|
encoder->SetBitrate(maxBitrate);
|
|
}
|
|
}else if(request==2){ // set packet loss
|
|
if(encoder){
|
|
encoder->SetPacketLoss(param);
|
|
}
|
|
}else if(request==3){ // force enable/disable p2p
|
|
allowP2p=param==1;
|
|
if(!allowP2p && currentEndpoint && currentEndpoint->type!=EP_TYPE_UDP_RELAY){
|
|
currentEndpoint=preferredRelay;
|
|
}else if(allowP2p){
|
|
SendPublicEndpointsRequest();
|
|
}
|
|
BufferOutputStream s(4);
|
|
s.WriteInt32(dataSavingMode ? INIT_FLAG_DATA_SAVING_ENABLED : 0);
|
|
SendPacketReliably(PKT_NETWORK_CHANGED, s.GetBuffer(), s.GetLength(), 1, 20);
|
|
}else if(request==4){
|
|
if(echoCanceller)
|
|
echoCanceller->Enable(param==1);
|
|
}
|
|
}
|
|
|
|
|
|
const char* VoIPController::GetVersion(){
|
|
return LIBTGVOIP_VERSION;
|
|
}
|
|
|
|
|
|
int64_t VoIPController::GetPreferredRelayID(){
|
|
if(preferredRelay)
|
|
return preferredRelay->id;
|
|
return 0;
|
|
}
|
|
|
|
|
|
int VoIPController::GetLastError(){
|
|
return lastError;
|
|
}
|
|
|
|
|
|
void VoIPController::GetStats(voip_stats_t *stats){
|
|
memcpy(stats, &this->stats, sizeof(voip_stats_t));
|
|
}
|
|
|
|
#ifdef TGVOIP_USE_AUDIO_SESSION
|
|
void VoIPController::SetAcquireAudioSession(void (^completion)(void (^)())) {
|
|
this->acquireAudioSession = [completion copy];
|
|
}
|
|
|
|
void VoIPController::ReleaseAudioSession(void (^completion)()) {
|
|
completion();
|
|
}
|
|
#endif
|
|
|
|
void VoIPController::LogDebugInfo(){
|
|
std::string json="{\"endpoints\":[";
|
|
for(std::vector<Endpoint*>::iterator itr=endpoints.begin();itr!=endpoints.end();++itr){
|
|
Endpoint* e=*itr;
|
|
char buffer[1024];
|
|
const char* typeStr="unknown";
|
|
switch(e->type){
|
|
case EP_TYPE_UDP_RELAY:
|
|
typeStr="udp_relay";
|
|
break;
|
|
case EP_TYPE_UDP_P2P_INET:
|
|
typeStr="udp_p2p_inet";
|
|
break;
|
|
case EP_TYPE_UDP_P2P_LAN:
|
|
typeStr="udp_p2p_lan";
|
|
break;
|
|
case EP_TYPE_TCP_RELAY:
|
|
typeStr="tcp_relay";
|
|
break;
|
|
}
|
|
snprintf(buffer, 1024, "{\"address\":\"%s\",\"port\":%u,\"type\":\"%s\",\"rtt\":%u%s%s}", e->address.ToString().c_str(), e->port, typeStr, (unsigned int)round(e->averageRTT*1000), currentEndpoint==&*e ? ",\"in_use\":true" : "", preferredRelay==&*e ? ",\"preferred\":true" : "");
|
|
json+=buffer;
|
|
if(itr!=endpoints.end()-1)
|
|
json+=",";
|
|
}
|
|
json+="],";
|
|
char buffer[1024];
|
|
const char* netTypeStr;
|
|
switch(networkType){
|
|
case NET_TYPE_WIFI:
|
|
netTypeStr="wifi";
|
|
break;
|
|
case NET_TYPE_GPRS:
|
|
netTypeStr="gprs";
|
|
break;
|
|
case NET_TYPE_EDGE:
|
|
netTypeStr="edge";
|
|
break;
|
|
case NET_TYPE_3G:
|
|
netTypeStr="3g";
|
|
break;
|
|
case NET_TYPE_HSPA:
|
|
netTypeStr="hspa";
|
|
break;
|
|
case NET_TYPE_LTE:
|
|
netTypeStr="lte";
|
|
break;
|
|
case NET_TYPE_ETHERNET:
|
|
netTypeStr="ethernet";
|
|
break;
|
|
case NET_TYPE_OTHER_HIGH_SPEED:
|
|
netTypeStr="other_high_speed";
|
|
break;
|
|
case NET_TYPE_OTHER_LOW_SPEED:
|
|
netTypeStr="other_low_speed";
|
|
break;
|
|
case NET_TYPE_DIALUP:
|
|
netTypeStr="dialup";
|
|
break;
|
|
case NET_TYPE_OTHER_MOBILE:
|
|
netTypeStr="other_mobile";
|
|
break;
|
|
default:
|
|
netTypeStr="unknown";
|
|
break;
|
|
}
|
|
snprintf(buffer, 1024, "\"time\":%u,\"network_type\":\"%s\"}", (unsigned int)time(NULL), netTypeStr);
|
|
json+=buffer;
|
|
debugLogs.push_back(json);
|
|
}
|
|
|
|
std::string VoIPController::GetDebugLog(){
|
|
std::string log="{\"events\":[";
|
|
|
|
for(std::vector<std::string>::iterator itr=debugLogs.begin();itr!=debugLogs.end();++itr){
|
|
log+=(*itr);
|
|
if((itr+1)!=debugLogs.end())
|
|
log+=",";
|
|
}
|
|
log+="],\"libtgvoip_version\":\"" LIBTGVOIP_VERSION "\"}";
|
|
return log;
|
|
}
|
|
|
|
void VoIPController::GetDebugLog(char *buffer){
|
|
strcpy(buffer, GetDebugLog().c_str());
|
|
}
|
|
|
|
size_t VoIPController::GetDebugLogLength(){
|
|
size_t len=128;
|
|
for(std::vector<std::string>::iterator itr=debugLogs.begin();itr!=debugLogs.end();++itr){
|
|
len+=(*itr).length()+1;
|
|
}
|
|
return len;
|
|
}
|
|
|
|
std::vector<AudioInputDevice> VoIPController::EnumerateAudioInputs(){
|
|
vector<AudioInputDevice> devs;
|
|
audio::AudioInput::EnumerateDevices(devs);
|
|
return devs;
|
|
}
|
|
|
|
std::vector<AudioOutputDevice> VoIPController::EnumerateAudioOutputs(){
|
|
vector<AudioOutputDevice> devs;
|
|
audio::AudioOutput::EnumerateDevices(devs);
|
|
return devs;
|
|
}
|
|
|
|
void VoIPController::SetCurrentAudioInput(std::string id){
|
|
currentAudioInput=id;
|
|
if(audioInput)
|
|
audioInput->SetCurrentDevice(id);
|
|
}
|
|
|
|
void VoIPController::SetCurrentAudioOutput(std::string id){
|
|
currentAudioOutput=id;
|
|
if(audioOutput)
|
|
audioOutput->SetCurrentDevice(id);
|
|
}
|
|
|
|
std::string VoIPController::GetCurrentAudioInputID(){
|
|
return currentAudioInput;
|
|
}
|
|
|
|
std::string VoIPController::GetCurrentAudioOutputID(){
|
|
return currentAudioOutput;
|
|
}
|
|
|
|
void VoIPController::SetProxy(int protocol, std::string address, uint16_t port, std::string username, std::string password){
|
|
proxyProtocol=protocol;
|
|
proxyAddress=address;
|
|
proxyPort=port;
|
|
proxyUsername=username;
|
|
proxyPassword=password;
|
|
}
|
|
|
|
void VoIPController::SendUdpPing(Endpoint *endpoint){
|
|
if(endpoint->type!=EP_TYPE_UDP_RELAY)
|
|
return;
|
|
LOGV("Sending UDP ping to %s:%d", endpoint->address.ToString().c_str(), endpoint->port);
|
|
BufferOutputStream p(1024);
|
|
p.WriteBytes(endpoint->peerTag, 16);
|
|
p.WriteInt32(-1);
|
|
p.WriteInt32(-1);
|
|
p.WriteInt32(-1);
|
|
p.WriteInt32(-2);
|
|
p.WriteInt64(12345);
|
|
NetworkPacket pkt;
|
|
pkt.address=&endpoint->address;
|
|
pkt.port=endpoint->port;
|
|
pkt.protocol=PROTO_UDP;
|
|
pkt.data=p.GetBuffer();
|
|
pkt.length=p.GetLength();
|
|
udpSocket->Send(&pkt);
|
|
}
|
|
|
|
int VoIPController::GetSignalBarsCount(){
|
|
unsigned char avg=0;
|
|
for(int i=0;i<sizeof(signalBarsHistory);i++)
|
|
avg+=signalBarsHistory[i];
|
|
return avg >> 2;
|
|
}
|
|
|
|
void VoIPController::SetSignalBarsCountCallback(void (*f)(VoIPController *, int)){
|
|
signalBarCountCallback=f;
|
|
}
|
|
|
|
void VoIPController::SetAudioOutputGainControlEnabled(bool enabled){
|
|
LOGD("New output AGC state: %d", enabled);
|
|
outputAGCEnabled=enabled;
|
|
if(outputAGC)
|
|
outputAGC->SetPassThrough(!enabled);
|
|
}
|
|
|
|
void VoIPController::SetEchoCancellationStrength(int strength){
|
|
echoCancellationStrength=strength;
|
|
if(echoCanceller)
|
|
echoCanceller->SetAECStrength(strength);
|
|
}
|
|
|
|
Endpoint::Endpoint(int64_t id, uint16_t port, IPv4Address& _address, IPv6Address& _v6address, char type, unsigned char peerTag[16]) : address(_address), v6address(_v6address){
|
|
this->id=id;
|
|
this->port=port;
|
|
this->type=type;
|
|
memcpy(this->peerTag, peerTag, 16);
|
|
LOGV("new endpoint %lld: %s:%u", (long long int)id, address.ToString().c_str(), port);
|
|
|
|
lastPingSeq=0;
|
|
lastPingTime=0;
|
|
averageRTT=0;
|
|
memset(rtts, 0, sizeof(rtts));
|
|
socket=NULL;
|
|
}
|
|
|
|
Endpoint::Endpoint() : address(0), v6address("::0") {
|
|
lastPingSeq=0;
|
|
lastPingTime=0;
|
|
averageRTT=0;
|
|
memset(rtts, 0, sizeof(rtts));
|
|
socket=NULL;
|
|
}
|
|
|
|
#if defined(__APPLE__) && TARGET_OS_IPHONE
|
|
void VoIPController::SetRemoteEndpoints(voip_legacy_endpoint_t* buffer, size_t count, bool allowP2P){
|
|
std::vector<Endpoint> endpoints;
|
|
for(size_t i=0;i<count;i++){
|
|
voip_legacy_endpoint_t e=buffer[i];
|
|
IPv4Address v4addr=IPv4Address(std::string(e.address));
|
|
IPv6Address v6addr=IPv6Address(std::string(e.address6));
|
|
endpoints.push_back(Endpoint(e.id, e.port, v4addr, v6addr, EP_TYPE_UDP_RELAY, e.peerTag));
|
|
}
|
|
this->SetRemoteEndpoints(endpoints, allowP2P);
|
|
}
|
|
#endif
|