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rtp.cpp
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rtp.cpp
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/* Martin Vit [email protected]
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2.
*/
/*
This unit implements class RTP which processes RTP packets and make statistics on them.
Each Call class contains two RTP classes.
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <sys/types.h>
#include <syslog.h>
#include <errno.h>
#include <pcap.h>
#include "tools.h"
#include "rtp.h"
#include "calltable.h"
#include "codecs.h"
#include "sniff.h"
#include "jitterbuffer/asterisk/channel.h"
#include "jitterbuffer/asterisk/frame.h"
#include "jitterbuffer/asterisk/abstract_jb.h"
#include "jitterbuffer/asterisk/strings.h"
extern int verbosity;
extern int opt_saveRAW; //save RTP payload RAW data?
extern int opt_saveWAV; //save RTP payload RAW data?
extern int opt_saveGRAPH; //save GRAPH data?
extern int opt_gzipGRAPH; //save gzip GRAPH data?
extern int opt_jitterbuffer_f1; // turns off/on jitterbuffer simulator to compute MOS score mos_f1
extern int opt_jitterbuffer_f2; // turns off/on jitterbuffer simulator to compute MOS score mos_f2
extern int opt_jitterbuffer_adapt; // turns off/on jitterbuffer simulator to compute MOS score mos_adapt
extern char opt_cachedir[1024];
extern int opt_savewav_force;
int dtmfdebug = 0;
using namespace std;
/* Convert timeval structure into microsecond representation */
inline u_int32_t timeval2micro(const timeval t) {
return ((t.tv_sec * 1000000ul) + t.tv_usec);
}
/* Subtract the `struct timeval' values X and Y,
storing the result in RESULT.
Return 1 if the difference is negative, otherwise 0. */
int
timeval_subtract (struct timeval *result, struct timeval x, struct timeval y) {
/* Perform the carry for the later subtraction by updating y. */
if (x.tv_usec < y.tv_usec) {
int nsec = (y.tv_usec - x.tv_usec) / 1000000 + 1;
y.tv_usec -= 1000000 * nsec;
y.tv_sec += nsec;
}
if (x.tv_usec - y.tv_usec > 1000000) {
int nsec = (x.tv_usec - y.tv_usec) / 1000000;
y.tv_usec += 1000000 * nsec;
y.tv_sec -= nsec;
}
/* Compute the time remaining to wait.
tv_usec is certainly p60itive. */
result->tv_sec = x.tv_sec - y.tv_sec;
result->tv_usec = x.tv_usec - y.tv_usec;
/* Return 1 if result is negative. */
return x.tv_sec < y.tv_sec;
}
int get_ticks_bycodec(int codec) {
switch(codec) {
case PAYLOAD_PCMU:
return 8;
break;
case PAYLOAD_GSM:
return 8;
break;
case PAYLOAD_G723:
return 8;
break;
case PAYLOAD_PCMA:
return 8;
break;
case PAYLOAD_G722:
return 8;
break;
case PAYLOAD_G729:
return 8;
break;
case PAYLOAD_ILBC:
return 8;
break;
case PAYLOAD_SPEEX:
return 8;
break;
case PAYLOAD_SILK8:
return 8;
break;
case PAYLOAD_SILK12:
return 12;
break;
case PAYLOAD_SILK16:
return 16;
break;
case PAYLOAD_SILK24:
return 24;
break;
case PAYLOAD_ISAC16:
return 16;
break;
case PAYLOAD_ISAC32:
return 32;
break;
case PAYLOAD_OPUS:
return 8;
break;
case PAYLOAD_OPUS12:
return 12;
break;
case PAYLOAD_OPUS16:
return 16;
break;
case PAYLOAD_OPUS24:
return 24;
break;
case PAYLOAD_OPUS48:
return 48;
break;
default:
return 8;
}
}
/* constructor */
RTP::RTP() {
samplerate = 8000;
first = true;
first_packet_time = 0;
first_packet_usec = 0;
s = new source;
memset(s, 0, sizeof(source));
memset(&stats, 0, sizeof(stats));
memset(&rtcp, 0, sizeof(rtcp));
nintervals = 1;
saddr = 0;
daddr = 0;
ssrc = 0;
ssrc2 = 0;
gfilename[0] = '\0';
gfileRAW = NULL;
channel_fix1 = (ast_channel*)calloc(1, sizeof(*channel_fix1));
channel_fix1->jitter_impl = 0; // fixed
channel_fix1->jitter_max = 60;
channel_fix1->jitter_resync_threshold = 50;
channel_fix1->last_datalen = 0;
channel_fix1->lastbuflen = 0;
channel_fix1->resync = 1;
channel_fix1->audiobuf = NULL;
channel_fix2 = (ast_channel*)calloc(1, sizeof(*channel_fix2));
channel_fix2->jitter_impl = 0; // fixed
channel_fix2->jitter_max = 200;
channel_fix2->jitter_resync_threshold = 200;
channel_fix2->last_datalen = 0;
channel_fix2->lastbuflen = 0;
channel_fix2->resync = 1;
channel_fix2->audiobuf = NULL;
channel_adapt = (ast_channel*)calloc(1, sizeof(*channel_adapt));
channel_adapt->jitter_impl = 1; // adaptive
channel_adapt->jitter_max = 500;
channel_adapt->jitter_resync_threshold = 500;
channel_adapt->last_datalen = 0;
channel_adapt->lastbuflen = 0;
channel_adapt->resync = 1;
channel_adapt->audiobuf = NULL;
channel_record = (ast_channel*)calloc(1, sizeof(*channel_record));
channel_record->jitter_impl = 0; // fixed
channel_record->jitter_max = 60;
channel_record->jitter_resync_threshold = 1000;
channel_record->last_datalen = 0;
channel_record->lastbuflen = 0;
channel_record->resync = 0;
channel_record->audiobuf = NULL;
//channel->name = "SIP/fixed";
frame = (ast_frame*)calloc(1, sizeof(*frame));
frame->frametype = AST_FRAME_VOICE;
lastframetype = AST_FRAME_VOICE;
//frame->src = "DUMMY";
last_seq = 0;
last_ts = 0;
packetization = 0;
last_packetization = 0;
packetization_iterator = 0;
payload = -1;
prev_payload = -1;
payload2 = -1;
codec = -1;
for(int i = 0; i < MAX_RTPMAP; i++) {
rtpmap[i] = 0;
}
gfileRAW_buffer = NULL;
sid = false;
prev_sid = false;
call_owner = NULL;
pinformed = 0;
last_end_timestamp = 0;
lastdtmf = 0;
}
/* destructor */
RTP::~RTP() {
/*
if(packetization)
RTP::dump();
*/
Call *owner = (Call*)call_owner;
if(verbosity > 9) {
RTP::dump();
}
if(gfileRAW) {
jitterbuffer_fixed_flush(channel_record);
fclose(gfileRAW);
}
delete s;
ast_jb_destroy(channel_fix1);
ast_jb_destroy(channel_fix2);
ast_jb_destroy(channel_adapt);
ast_jb_destroy(channel_record);
free(channel_fix1);
free(channel_fix2);
free(channel_adapt);
free(channel_record);
free(frame);
if(opt_saveGRAPH || (owner && (owner->flags & FLAG_SAVEGRAPH))) {
if(opt_gzipGRAPH && gfileGZ.is_open()) {
gfileGZ.close();
} else if(gfile.is_open()){
gfile.close();
}
if(gfilename[0] != '\0') {
if(opt_cachedir[0] != '\0') {
owner->addtocachequeue(string(gfilename));
}
owner->addtofilesqueue(string(gfilename), "graphsize");
}
}
if(gfileRAW_buffer) {
free(gfileRAW_buffer);
}
}
const unsigned int RTP::get_payload_len() {
payload_data = data + sizeof(RTPFixedHeader);
payload_len = len - sizeof(RTPFixedHeader);
if(getPadding()) {
/*
* If set, this packet contains one or more additional padding
* bytes at the end which are not part of the payload. The last
* byte of the padding contains a count of how many padding bytes
* should be ignored. Padding may be needed by some encryption
* algorithms with fixed block sizes or for carrying several RTP
* packets in a lower-layer protocol data unit.
*/
payload_len -= ((u_int8_t *)data)[payload_len - 1];
}
if(getCC() > 0) {
/*
* The number of CSRC identifiers that follow the fixed header.
*/
payload_data += 4 * getCC();
payload_len -= 4 * getCC();
}
if(getExtension()) {
/*
* If set, the fixed header is followed by exactly one header extension.
*/
extension_hdr_t *rtpext;
if (payload_len < 4)
payload_len = 0;
// the extension, if present, is after the CSRC list.
rtpext = (extension_hdr_t *)((u_int8_t *)payload_data);
payload_data += sizeof(extension_hdr_t) + rtpext->length;
payload_len -= sizeof(extension_hdr_t) + rtpext->length;
}
return payload_len;
}
/* flush jitterbuffer */
void RTP::jitterbuffer_fixed_flush(struct ast_channel *jchannel) {
jb_fixed_flush_deliver(channel_record);
}
/* add silence to RTP stream from last packet time to current time which is in header->ts */
void
RTP::jt_tail(struct pcap_pkthdr *header) {
if(!ast_jb_test(channel_record)) {
// there is no ongoing recording, return
return;
}
/* protect for endless loops (it cannot happen in theory but to be sure */
if(packetization <= 0) {
Call *owner = (Call*)call_owner;
if(owner) {
syslog(LOG_ERR, "call-id[%s]: packetization is 0 in jitterbuffer function.", owner->get_fbasename_safe());
} else {
syslog(LOG_ERR, "call-id[N/A]: packetization is 0 in jitterbuffer function.");
}
return;
}
/* calculate time difference between last pakcet and current packet + packetization time*/
if(channel_record->last_ts.tv_sec == 0) {
// previouuse tv_sec is not set, set it
memcpy(&channel_record->last_ts, &header->ts, sizeof(timeval));
return;
}
int msdiff = ast_tvdiff_ms(header->ts, channel_record->last_ts);
msdiff -= packetization;
while( msdiff >= packetization ) {
ast_jb_get_and_deliver(channel_record, &channel_record->last_ts);
/* adding packetization time to last_ts time */
struct timeval tmp = ast_tvadd(channel_record->last_ts, ast_samp2tv(packetization, 1000));
memcpy(&channel_record->last_ts, &tmp, sizeof(struct timeval));
msdiff -= packetization;
}
}
#if 1
/* simulate jitterbuffer */
void
RTP::jitterbuffer(struct ast_channel *channel, int savePayload) {
Call *owner = (Call*)call_owner;
if(owner and savePayload and owner->silencerecording) {
// skip recording
frame->skip = 1;
} else {
frame->skip = 0;
}
struct timeval tsdiff;
switch(codec) {
case PAYLOAD_OPUS12:
frame->ts = getTimestamp() / 12;
frame->len = packetization * 2 / 3;
break;
case PAYLOAD_ISAC16:
case PAYLOAD_SILK16:
case PAYLOAD_OPUS16:
frame->ts = getTimestamp() / 16;
frame->len = packetization / 2;
break;
case PAYLOAD_SILK24:
case PAYLOAD_OPUS24:
frame->ts = getTimestamp() / 24;
frame->len = packetization / 3;
break;
case PAYLOAD_OPUS48:
frame->ts = getTimestamp() / 48;
frame->len = packetization / 6;
break;
default:
frame->ts = getTimestamp() / 8;
frame->len = packetization;
}
frame->marker = getMarker();
frame->seqno = getSeqNum();
channel->codec = codec;
memcpy(&frame->delivery, &header->ts, sizeof(struct timeval));
/* protect for endless loops (it cannot happen in theory but to be sure */
if(packetization <= 0) {
if(pinformed == 0) {
if(owner) {
syslog(LOG_ERR, "call-id[%s] ssrc[%x]: packetization is 0 in jitterbuffer function.", owner->get_fbasename_safe(), getSSRC());
} else {
syslog(LOG_ERR, "call-id[N/A] ssrc[%x]: packetization is 0 in jitterbuffer function.", getSSRC());
}
}
pinformed = 1;
return;
} else {
pinformed = 0;
}
if(savePayload) {
struct iphdr2 *header_ip = (struct iphdr2 *)(data - sizeof(struct iphdr2) - sizeof(udphdr2));
int mylen = MIN(len, ntohs(header_ip->tot_len) - header_ip->ihl * 4 - sizeof(udphdr2));
/* get RTP payload header and datalen */
payload_data = data + sizeof(RTPFixedHeader);
payload_len = mylen - sizeof(RTPFixedHeader);
if(getPadding()) {
/*
* If set, this packet contains one or more additional padding
* bytes at the end which are not part of the payload. The last
* byte of the padding contains a count of how many padding bytes
* should be ignored. Padding may be needed by some encryption
* algorithms with fixed block sizes or for carrying several RTP
* packets in a lower-layer protocol data unit.
*/
payload_len -= ((u_int8_t *)data)[payload_len - 1];
}
if(getCC() > 0) {
/*
* The number of CSRC identifiers that follow the fixed header.
*/
payload_data += 4 * getCC();
payload_len -= 4 * getCC();
}
if(getExtension()) {
/*
* If set, the fixed header is followed by exactly one header extension.
*/
extension_hdr_t *rtpext;
if (payload_len < 4)
payload_len = 0;
// the extension, if present, is after the CSRC list.
rtpext = (extension_hdr_t *)((u_int8_t *)payload_data);
payload_data += sizeof(extension_hdr_t) + rtpext->length;
payload_len -= sizeof(extension_hdr_t) + rtpext->length;
}
frame->data = payload_data;
frame->datalen = payload_len > 0 ? payload_len : 0; /* ensure that datalen is never negative */
if(getPayload() == PAYLOAD_G723) {
// voipmonitor does not handle SID packets well (silence packets) it causes out of sync
if((unsigned char)payload_data[0] & 2) {
/* check if jitterbuffer is already created. If not we have to create it because
if call starts with SID packets first it will than cause out of sync calls
*/
if(ast_test_flag(&channel->jb, (1 << 2))) {
// jitterbuffer is created so we can skip SID packets now
return;
}
}
}
channel->rawstream = gfileRAW;
Call *owner = (Call*)call_owner;
if(iscaller) {
owner->codec_caller = codec;
if(owner->audiobuffer1) {
channel->audiobuf = owner->audiobuffer1;
}
} else {
owner->codec_called = codec;
if(owner->audiobuffer2) {
channel->audiobuf = owner->audiobuffer2;
}
}
if(payload_len > 0) {
channel->last_datalen = frame->datalen;
}
} else {
frame->datalen = 0;
frame->data = NULL;
channel->rawstream = NULL;
}
// create jitter buffer structures
ast_jb_do_usecheck(channel, &header->ts);
if(!channel->jb_reseted) {
// initializing jitterbuffer
if(savePayload) {
channel_record->jitter_max = frame->len * 3;
}
ast_jb_empty_and_reset(channel);
channel->jb_reseted = 1;
memcpy(&channel->last_ts, &header->ts, sizeof(struct timeval));
ast_jb_put(channel, frame, &header->ts);
return;
}
/* calculate time difference between last pakcet and current packet + packetization time*/
int msdiff = ast_tvdiff_ms( header->ts, ast_tvadd(channel->last_ts, ast_samp2tv(packetization, 1000)) );
//printf("ms:%d\n", msdiff);
if(msdiff > packetization * 1000) {
// difference is too big, reseting last_ts to current packet. If we dont check this it could happen to run while cycle endlessly
memcpy(&channel->last_ts, &header->ts, sizeof(struct timeval));
ast_jb_put(channel, frame, &header->ts);
if(verbosity > 4) syslog(LOG_ERR, "big timestamp jump (msdiff:%d packetization: %d) in this file: %s\n", msdiff, packetization, gfilename);
return;
}
/* between last packet and current packet is big timestamp difference and it could count
* interpolated framed although it was silence so calculate real number of packets based
* on timestamps in packet header, timestamps in rtp header and sequence numbers between
* last packet and current packet
*/
// relative time difference calculated from packet sequence
u_int32_t sequencems = (frame->seqno - last_seq) * packetization;
/* difference (in ms) between timestamps in packet header and rtp timestamps. this should
* be ideally equel to zero. Negative values mean that packet arrives earlier and positive
* values indicates that packet was late
*/
long double transit = (timeval_subtract(&tsdiff, header->ts, s->lastTimeRec) ? -timeval2micro(tsdiff)/1000.0 : timeval2micro(tsdiff)/1000.0) - (double)(getTimestamp() - s->lastTimeStamp)/(double)samplerate/1000;
/* and now if there is bigger (lets say one second) timestamp difference (calculated from pakcet headers)
* between two last packets and transit time is equel or smaller than sequencems (with 200ms toleration),
* it was silence and manually mark the frame which indicates to not count interpolated frame and resynchronize jitterbuffer
*/
if( msdiff > 1000 and (transit <= (sequencems + 200)) ) {
if(verbosity > 4) printf("jitterbuffer: manually marking packet, msdiff(%d) > 1000 and transit (%Lf) <= ((sequencems(%u) + 200)\n", msdiff, transit, sequencems);
frame->marker = 1;
}
// fetch pakcet from jitterbuffer every 20 ms regardless on packet loss or delay
while( msdiff >= packetization ) {
if(frame->marker or lastframetype == AST_FRAME_DTMF) {
/* if last frame was marked or DTMF, ignore interpolated frames */
channel->last_loss_burst = 0;
}
ast_jb_get_and_deliver(channel, &channel->last_ts);
/* adding packetization time to last_ts time */
struct timeval tmp = ast_tvadd(channel->last_ts, ast_samp2tv(frame->len, 1000));
memcpy(&channel->last_ts, &tmp, sizeof(struct timeval));
msdiff -= packetization;
}
ast_jb_put(channel, frame, &header->ts);
}
#endif
void
RTP::process_dtmf_rfc2833() {
unsigned int seqno = getSeqNum();
unsigned int event, event_end, samples;
char resp = 0;
unsigned int timestamp = getTimestamp();
unsigned char *pdata = data + sizeof(RTPFixedHeader);
/* Figure out event, event end, and samples */
event = ntohl(*((unsigned int *)(pdata)));
event >>= 24;
event_end = ntohl(*((unsigned int *)(pdata)));
event_end <<= 8;
event_end >>= 24;
samples = ntohl(*((unsigned int *)(pdata)));
samples &= 0xFFFF;
if(dtmfdebug) syslog(LOG_ERR, "Got RTP RFC2833 from %u (seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d) \n",
getMarker(), seqno, timestamp, len, (getMarker()?1:0), event, ((event_end & 0x80)?1:0), samples);
/* Figure out what digit was pressed */
if (event < 10) {
resp = '0' + event;
} else if (event < 11) {
resp = '*';
} else if (event < 12) {
resp = '#';
} else if (event < 16) {
resp = 'A' + (event - 12);
} else if (event < 17) { /* Event 16: Hook flash */
resp = 'X';
} else {
/* Not a supported event */
//syslog(LOG_ERR, "Ignoring RTP 2833 Event: %08x. Not a DTMF Digit.\n", event);
return;
}
if ((last_end_timestamp != timestamp) || (lastdtmf && lastdtmf != resp)) {
lastdtmf = resp;
if(dtmfdebug) syslog(LOG_ERR, "dtmfevent %c\n", resp);
last_end_timestamp = timestamp;
Call *owner = (Call*)call_owner;
if(owner) {
owner->handle_dtmf(resp, ts2double(header->ts.tv_sec, header->ts.tv_usec), saddr, daddr);
}
}
return;
}
/* read rtp packet */
void
RTP::read(unsigned char* data, int len, struct pcap_pkthdr *header, u_int32_t saddr, u_int32_t daddr, int seeninviteok) {
this->data = data;
this->len = len;
this->header = header;
this->saddr = saddr;
this->daddr = daddr;
if(this->first_packet_time == 0 and this->first_packet_usec == 0) {
this->first_packet_time = header->ts.tv_sec;
this->first_packet_usec = header->ts.tv_usec;
}
Call *owner = (Call*)call_owner;
// if(getSSRC() != 0x84a4eeea) return;
if(getVersion() != 2) {
return;
}
u_int16_t seq = getSeqNum();
if(seq == last_seq) {
// ignore duplicated RTP packets
return;
}
int curpayload = getPayload();
// printf("p[%d]\n", curpayload);
// ignore CNG
if(curpayload == 13 or curpayload == 19) {
last_seq = seq;
if(update_seq(seq)) {
update_stats();
}
return;
}
if(!owner) return;
if(curpayload == 101) {
process_dtmf_rfc2833();
}
/* codec changed */
if((codec == -1 || (curpayload != prev_payload)) && (curpayload != 101 && prev_payload != 101)) {
if(curpayload >= 96 && curpayload <= 127) {
/* for dynamic payload we look into rtpmap */
for(int i = 0; i < MAX_RTPMAP; i++) {
if(rtpmap[i] != 0 && curpayload == rtpmap[i] / 1000) {
codec = rtpmap[i] - curpayload * 1000;
}
}
} else {
codec = curpayload;
}
switch(codec) {
case PAYLOAD_SILK12:
case PAYLOAD_OPUS12:
samplerate = 12000;
break;
case PAYLOAD_ISAC16:
case PAYLOAD_SILK16:
case PAYLOAD_OPUS16:
samplerate = 16000;
break;
case PAYLOAD_SILK24:
case PAYLOAD_OPUS24:
samplerate = 24000;
break;
case PAYLOAD_ISAC32:
samplerate = 32000;
break;
case PAYLOAD_OPUS48:
samplerate = 48000;
break;
default:
samplerate = 8000;
}
if(iscaller) {
owner->last_callercodec = codec;
} else {
owner->last_calledcodec = codec;
}
if(opt_saveRAW || opt_savewav_force || (owner && (owner->flags & FLAG_SAVEWAV)) ||
(owner && (owner->audiobuffer1 || owner->audiobuffer2))// if recording requested
) {
// if(verbosity > 0) syslog(LOG_ERR, "converting WAV! [%u]\n", owner->flags);
/* open file for raw codec */
unsigned long unique = getTimestamp();
char tmp[1024];
sprintf(tmp, "%s.%d.%lu.%d.%ld.%ld.raw", basefilename, ssrc_index, unique, codec, header->ts.tv_sec, header->ts.tv_usec);
if(gfileRAW) {
//there is already opened gfileRAW
jitterbuffer_fixed_flush(channel_record);
fclose(gfileRAW);
} else {
/* look for the last RTP stream belonging to this direction and let jitterbuffer put silence
* which fills potentionally gap between this and previouse RTP so it will stay in sync with
* the other direction of call
*/
RTP *prevrtp = (RTP*)(owner->rtp_prev[iscaller]);
if(prevrtp && prevrtp != this) {
prevrtp->data = data;
prevrtp->len = len;
prevrtp->header = header;
prevrtp->saddr = saddr;
prevrtp->daddr = daddr;
prevrtp->jitterbuffer(prevrtp->channel_record, opt_saveRAW || opt_savewav_force || (owner->flags & FLAG_SAVEWAV) || (owner && (owner->audiobuffer1 || owner->audiobuffer2)));
}
}
gfileRAW = fopen(tmp, "w");
if(!gfileRAW_buffer) {
gfileRAW_buffer = (char*)malloc(32768 * sizeof(char));
if(gfileRAW_buffer == NULL) {
syslog(LOG_ERR, "Cannot allocate memory for gfileRAW_buffer - low memory this is FATAL");
exit(2);
}
}
if(!gfileRAW) {
syslog(LOG_ERR, "Cannot open file %s for writing: %s\n", tmp, strerror (errno));
exit(2);
}
if(gfileRAW_buffer) {
setvbuf(gfileRAW, gfileRAW_buffer, _IOFBF, 32768);
}
/* write file info to "playlist" */
sprintf(tmp, "%s.rawInfo", basefilename);
FILE *gfileRAWInfo = fopen(tmp, "a");
if(gfileRAWInfo) {
fprintf(gfileRAWInfo, "%d:%lu:%d:%ld:%ld\n", ssrc_index, unique, codec, header->ts.tv_sec, header->ts.tv_usec);
fclose(gfileRAWInfo);
} else {
syslog(LOG_ERR, "Cannot open file %s.rawInfo for writing\n", basefilename);
}
}
}
if(payload < 0 && curpayload != 101) {
/* save payload to statistics based on first payload. TODO: what if payload is dynamically changing? */
payload = curpayload;
}
if(curpayload == 101) {
frame->frametype = AST_FRAME_DTMF;
} else {
frame->frametype = AST_FRAME_VOICE;
}
frame->lastframetype = (enum ast_frame_type)(lastframetype);
// voipmonitor now handles RTP streams including progress XXX: remove this comment if it will be confirmed stable enough
// if(seeninviteok) {
if(packetization_iterator == 0 || packetization_iterator == 1) {
// we dont know packetization yet. Behave differently n G723 codec
if(curpayload == PAYLOAD_G723) {
default_packetization = 30;
/* check if RTP packet is not Silence packet (SID). Silence packets can have different
packetization and if call starts with SID packets it will guess wrong packetization.
typical sitation is for 60ms packetization and 30ms SID packetization */
payload_data = data + sizeof(RTPFixedHeader);
sid = (unsigned char)payload_data[0] & 2;
} else {
sid = false;
default_packetization = 20;
}
}
if(packetization_iterator == 0) {
if(last_ts != 0 && seq == (last_seq + 1) && (prev_payload != 101 && curpayload != 101) && !sid && !prev_sid) {
// sequence numbers are ok, we can calculate packetization
if(curpayload == PAYLOAD_G729) {
// if G729 packet len is 20, packet len is 20ms. In other cases - will be added later (do not have 40ms packetizations samples right now)
if(get_payload_len() == 20) {
packetization = 20;
} else {
packetization = (getTimestamp() - last_ts) / 8;
}
} else {
packetization = (getTimestamp() - last_ts) / 8;
}
if(packetization > 0) {
last_packetization = packetization;
packetization_iterator++;
}
}
#if 1
// new way of getting packetization from packet datalen
if(curpayload == PAYLOAD_PCMU or curpayload == PAYLOAD_PCMA) {
channel_fix1->packetization = default_packetization =
channel_fix2->packetization = channel_adapt->packetization =
channel_record->packetization = packetization = get_payload_len() / 8;
if(packetization >= 10) {
if(verbosity > 3) printf("packetization:[%d] ssrc[%x]\n", packetization, getSSRC());
packetization_iterator = 10; // this will cause that packetization is estimated as final
if(opt_jitterbuffer_f1)
jitterbuffer(channel_fix1, 0);
if(opt_jitterbuffer_f2)
jitterbuffer(channel_fix2, 0);
if(opt_jitterbuffer_adapt)
jitterbuffer(channel_adapt, 0);
}
}
// new way of getting packetization from packet datalen
if(curpayload == PAYLOAD_GSM) {
channel_fix1->packetization = default_packetization =
channel_fix2->packetization = channel_adapt->packetization =
channel_record->packetization = packetization = get_payload_len() / 33 * 20;
if(packetization >= 10) {
if(verbosity > 3) printf("packetization:[%d] ssrc[%x]\n", packetization, getSSRC());
packetization_iterator = 10; // this will cause that packetization is estimated as final
if(opt_jitterbuffer_f1)
jitterbuffer(channel_fix1, 0);
if(opt_jitterbuffer_f2)
jitterbuffer(channel_fix2, 0);
if(opt_jitterbuffer_adapt)
jitterbuffer(channel_adapt, 0);
}
}
#endif
/* for recording, we cannot loose any packet */
if(opt_saveRAW || opt_savewav_force || (owner->flags & FLAG_SAVEWAV) || (owner && (owner->audiobuffer1 || owner->audiobuffer2))) { // if recording requested
if(packetization < 10) {
packetization = channel_record->packetization = default_packetization;
}
jitterbuffer(channel_record, opt_saveRAW || opt_savewav_force || (owner->flags & FLAG_SAVEWAV) || (owner && (owner->audiobuffer1 || owner->audiobuffer2)));
}
} else if(packetization_iterator == 1) {
if(last_ts != 0 && seq == (last_seq + 1) && curpayload != 101 && prev_payload != 101 && !sid && !prev_sid) {
// sequence numbers are ok, we can calculate packetization
if(curpayload == PAYLOAD_G729) {
// if G729 packet len is 20, packet len is 20ms. In other cases - will be added later (do not have 40ms packetizations samples right now)
if(get_payload_len() == 20) {
packetization = 20;
} else {
packetization = (getTimestamp() - last_ts) / 8;
}
} else {
packetization = (getTimestamp() - last_ts) / 8;
}
// now make packetization average
packetization = (packetization + last_packetization) / 2;
if(packetization <= 0 or getMarker()) {
// packetization failed or Marker bit is set, fall back to start
packetization_iterator = 0;
/* for recording, we cannot loose any packet */
if(opt_saveRAW || opt_savewav_force || (owner->flags & FLAG_SAVEWAV) ||
(owner && (owner->audiobuffer1 || owner->audiobuffer2))// if recording requested
){
packetization = channel_record->packetization = default_packetization;
jitterbuffer(channel_record, 1);
}
} else {
packetization_iterator++;
channel_fix1->packetization = channel_fix2->packetization = channel_adapt->packetization = channel_record->packetization = packetization;
if(verbosity > 3) printf("[%x] packetization:[%d]\n", getSSRC(), packetization);
if(opt_jitterbuffer_f1)
jitterbuffer(channel_fix1, 0);
if(opt_jitterbuffer_f2)
jitterbuffer(channel_fix2, 0);
if(opt_jitterbuffer_adapt)
jitterbuffer(channel_adapt, 0);
if(opt_saveRAW || opt_savewav_force || (owner->flags & FLAG_SAVEWAV) ||
(owner && (owner->audiobuffer1 || owner->audiobuffer2))// if recording requested
){
jitterbuffer(channel_record, 1);
}
}
} else {
packetization_iterator = 0;
/* for recording, we cannot loose any packet */
if(opt_saveRAW || opt_savewav_force || (owner->flags & FLAG_SAVEWAV) ||
(owner && (owner->audiobuffer1 || owner->audiobuffer2))// if recording requested
){
packetization = channel_record->packetization = default_packetization;
jitterbuffer(channel_record, 1);
}
}
} else {
if(last_ts != 0 and seq == (last_seq + 1) and curpayload != 101) {
// packetization can change over time
int curpacketization = (getTimestamp() - last_ts) / 8;
if(curpacketization % 10 == 0 and curpacketization >= 20 and curpacketization <= 120) {
channel_fix1->packetization = channel_fix2->packetization = channel_adapt->packetization = channel_record->packetization = packetization = curpacketization;
}
}
//printf("packetization [%d]\n", packetization);
if(opt_jitterbuffer_f1)
jitterbuffer(channel_fix1, 0);
if(opt_jitterbuffer_f2)
jitterbuffer(channel_fix2, 0);
if(opt_jitterbuffer_adapt)
jitterbuffer(channel_adapt, 0);
if(opt_saveRAW || opt_savewav_force || (owner->flags & FLAG_SAVEWAV) ||
(owner && (owner->audiobuffer1 || owner->audiobuffer2))// if recording requested
){
jitterbuffer(channel_record, 1);
}
}
// }
prev_payload = curpayload;
prev_sid = sid;
if(getMarker()) {
// if RTP packet is Marked, we have to reset last_ts to 0 so in next cycle it will count packetization from ground
last_ts = 0;
} else {
last_ts = getTimestamp();
}
last_seq = seq;
if(first) {
first = false;
init_seq(seq);
s->max_seq = seq - 1;
s->probation = MIN_SEQUENTIAL;
s->lastTimeRec = header->ts;
s->lastTimeStamp = getTimestamp();
} else {
if(update_seq(seq)) {
update_stats();
}
}
lastframetype = frame->frametype;
}
/* fill internal structures by the input RTP packet */
void
RTP::fill(unsigned char* data, int len, struct pcap_pkthdr *header, u_int32_t saddr, u_int32_t daddr) {
this->data = data;
this->len = len;
this->header = header;
this->saddr = saddr;
this->daddr = daddr;
}
/* update statistics data */
void
RTP::update_stats() {
int lost = int((s->cycles + s->max_seq - (s->base_seq + 1)) - s->received);
int adelay = 0;
struct timeval tsdiff;
double tsdiff2;
Call *owner = (Call*)call_owner;
/* if payload == 101 (EVENT) dont make delayes on this because it confuses stats */
if(getPayload() == 101)
return;
/* differences between last timestamp and current timestamp (timestamp from ip heade)
* frame1.time - frame0.time */
tsdiff2 = timeval_subtract(&tsdiff, header->ts, s->lastTimeRec) ? -timeval2micro(tsdiff)/1000.0 : timeval2micro(tsdiff)/1000.0;
long double transit = tsdiff2 - (double)(getTimestamp() - s->lastTimeStamp)/((double)samplerate/1000.0);
if(abs((int)transit) > 5000) {
/* timestamp skew, discard delay, it is possible that timestamp changed */
s->fdelay = s->avgdelay;
//s->fdelay = 0;
transit = 0;
} else {
adelay = abs(int(transit));
s->fdelay += transit;
}
/* Jitterbuffer calculation
* J(1) = J(0) + (|D(0,1)| - J(0))/16 */
if(transit < 0) transit = -transit;
double jitter = s->prevjitter + (double)(transit - s->prevjitter)/16. ;
s->prevjitter = jitter;
s->avgdelay = ((s->avgdelay * (long double)(s->received) - 1) + transit ) / (double)s->received;