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rtp.c
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rtp.c
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/*! \file rtp.c
* \author Lorenzo Miniero <[email protected]>
* \copyright GNU General Public License v3
* \brief RTP processing
* \details Implementation of the RTP header. Since the gateway does not
* much more than relaying frames around, the only thing we're interested
* in is the RTP header and how to get its payload, and parsing extensions.
*
* \ingroup protocols
* \ref protocols
*/
#include <string.h>
#include "rtp.h"
#include "debug.h"
char *janus_rtp_payload(char *buf, int len, int *plen) {
if(!buf || len < 12)
return NULL;
rtp_header *rtp = (rtp_header *)buf;
int hlen = 12;
if(rtp->csrccount) /* Skip CSRC if needed */
hlen += rtp->csrccount*4;
if(rtp->extension) {
janus_rtp_header_extension *ext = (janus_rtp_header_extension*)(buf+hlen);
int extlen = ntohs(ext->length)*4;
hlen += 4;
if(len > (hlen + extlen))
hlen += extlen;
}
if(plen)
*plen = len-hlen;
return buf+hlen;
}
int janus_rtp_header_extension_get_id(const char *sdp, const char *extension) {
if(!sdp || !extension)
return -1;
char extmap[100];
g_snprintf(extmap, 100, "a=extmap:%%d %s", extension);
/* Look for the extmap */
const char *line = strstr(sdp, "m=");
while(line) {
char *next = strchr(line, '\n');
if(next) {
*next = '\0';
if(strstr(line, "a=extmap") && strstr(line, extension)) {
/* Gotcha! */
int id = 0;
#pragma GCC diagnostic ignored "-Wformat-nonliteral"
if(sscanf(line, extmap, &id) == 1) {
#pragma GCC diagnostic warning "-Wformat-nonliteral"
*next = '\n';
return id;
}
}
*next = '\n';
}
line = next ? (next+1) : NULL;
}
return -2;
}
const char *janus_rtp_header_extension_get_from_id(const char *sdp, int id) {
if(!sdp || id < 0)
return NULL;
/* Look for the mapping */
char extmap[100];
g_snprintf(extmap, 100, "a=extmap:%d ", id);
const char *line = strstr(sdp, "m=");
while(line) {
char *next = strchr(line, '\n');
if(next) {
*next = '\0';
if(strstr(line, extmap)) {
/* Gotcha! */
char extension[100];
if(sscanf(line, "a=extmap:%d %s", &id, extension) == 2) {
*next = '\n';
if(strstr(extension, JANUS_RTP_EXTMAP_AUDIO_LEVEL))
return JANUS_RTP_EXTMAP_AUDIO_LEVEL;
if(strstr(extension, JANUS_RTP_EXTMAP_VIDEO_ORIENTATION))
return JANUS_RTP_EXTMAP_VIDEO_ORIENTATION;
if(strstr(extension, JANUS_RTP_EXTMAP_PLAYOUT_DELAY))
return JANUS_RTP_EXTMAP_PLAYOUT_DELAY;
if(strstr(extension, JANUS_RTP_EXTMAP_TOFFSET))
return JANUS_RTP_EXTMAP_TOFFSET;
if(strstr(extension, JANUS_RTP_EXTMAP_ABS_SEND_TIME))
return JANUS_RTP_EXTMAP_ABS_SEND_TIME;
if(strstr(extension, JANUS_RTP_EXTMAP_CC_EXTENSIONS))
return JANUS_RTP_EXTMAP_CC_EXTENSIONS;
if(strstr(extension, JANUS_RTP_EXTMAP_RTP_STREAM_ID))
return JANUS_RTP_EXTMAP_RTP_STREAM_ID;
JANUS_LOG(LOG_ERR, "Unsupported extension '%s'\n", extension);
return NULL;
}
}
*next = '\n';
}
line = next ? (next+1) : NULL;
}
return NULL;
}
/* Static helper to quickly find the extension data */
static int janus_rtp_header_extension_find(char *buf, int len, int id,
uint8_t *byte, uint32_t *word, char **ref) {
if(!buf || len < 12)
return -1;
rtp_header *rtp = (rtp_header *)buf;
int hlen = 12;
if(rtp->csrccount) /* Skip CSRC if needed */
hlen += rtp->csrccount*4;
if(rtp->extension) {
janus_rtp_header_extension *ext = (janus_rtp_header_extension *)(buf+hlen);
int extlen = ntohs(ext->length)*4;
hlen += 4;
if(len > (hlen + extlen)) {
/* 1-Byte extension */
if(ntohs(ext->type) == 0xBEDE) {
const uint8_t padding = 0x00, reserved = 0xF;
uint8_t extid = 0, idlen;
int i = 0;
while(i < extlen) {
extid = buf[hlen+i] >> 4;
if(extid == reserved) {
break;
} else if(extid == padding) {
i++;
continue;
}
idlen = (buf[hlen+i] & 0xF)+1;
if(extid == id) {
/* Found! */
if(byte)
*byte = buf[hlen+i+1];
if(word)
*word = *(uint32_t *)(buf+hlen+i);
if(ref)
*ref = &buf[hlen];
return 0;
}
i += 1 + idlen;
}
}
hlen += extlen;
}
}
return -1;
}
int janus_rtp_header_extension_parse_audio_level(char *buf, int len, int id, int *level) {
uint8_t byte = 0;
if(janus_rtp_header_extension_find(buf, len, id, &byte, NULL, NULL) < 0)
return -1;
/* a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level */
int v = (byte & 0x80) >> 7;
int value = byte & 0x7F;
JANUS_LOG(LOG_DBG, "%02x --> v=%d, level=%d\n", byte, v, value);
if(level)
*level = value;
return 0;
}
int janus_rtp_header_extension_parse_video_orientation(char *buf, int len, int id,
gboolean *c, gboolean *f, gboolean *r1, gboolean *r0) {
uint8_t byte = 0;
if(janus_rtp_header_extension_find(buf, len, id, &byte, NULL, NULL) < 0)
return -1;
/* a=extmap:4 urn:3gpp:video-orientation */
gboolean cbit = (byte & 0x08) >> 3;
gboolean fbit = (byte & 0x04) >> 2;
gboolean r1bit = (byte & 0x02) >> 1;
gboolean r0bit = byte & 0x01;
JANUS_LOG(LOG_DBG, "%02x --> c=%d, f=%d, r1=%d, r0=%d\n", byte, cbit, fbit, r1bit, r0bit);
if(c)
*c = cbit;
if(f)
*f = fbit;
if(r1)
*r1 = r1bit;
if(r0)
*r0 = r0bit;
return 0;
}
int janus_rtp_header_extension_parse_playout_delay(char *buf, int len, int id,
uint16_t *min_delay, uint16_t *max_delay) {
uint32_t bytes = 0;
if(janus_rtp_header_extension_find(buf, len, id, NULL, &bytes, NULL) < 0)
return -1;
/* a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay */
uint16_t min = (bytes & 0x00FFF000) >> 12;
uint16_t max = bytes & 0x00000FFF;
JANUS_LOG(LOG_DBG, "%"SCNu32"x --> min=%"SCNu16", max=%"SCNu16"\n", bytes, min, max);
if(min_delay)
*min_delay = min;
if(max_delay)
*max_delay = max;
return 0;
}
int janus_rtp_header_extension_parse_rtp_stream_id(char *buf, int len, int id,
char *sdes_item, int sdes_len) {
char *ext = NULL;
if(janus_rtp_header_extension_find(buf, len, id, NULL, NULL, &ext) < 0)
return -1;
/* a=extmap:3/sendonly urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id */
if(ext == NULL)
return -2;
int val_len = (*ext & 0x0F) + 1;
if(val_len > (sdes_len-1)) {
JANUS_LOG(LOG_WARN, "SDES buffer is too small (%d < %d), RTP stream ID will be cut\n", val_len, sdes_len);
val_len = sdes_len-1;
}
memcpy(sdes_item, ext+1, val_len);
*(sdes_item+val_len) = '\0';
return 0;
}
/* RTP context related methods */
void janus_rtp_switching_context_reset(janus_rtp_switching_context *context) {
if(context == NULL)
return;
/* Reset the context values */
memset(context, 0, sizeof(*context));
}
void janus_rtp_header_update(rtp_header *header, janus_rtp_switching_context *context, gboolean video, int step) {
if(header == NULL || context == NULL || step < 1)
return;
uint32_t ssrc = ntohl(header->ssrc);
uint32_t timestamp = ntohl(header->timestamp);
uint16_t seq = ntohs(header->seq_number);
if(video) {
if(ssrc != context->v_last_ssrc) {
/* Video SSRC changed: update both sequence number and timestamp */
JANUS_LOG(LOG_VERB, "Video SSRC changed, %"SCNu32" --> %"SCNu32"\n",
context->v_last_ssrc, ssrc);
context->v_last_ssrc = ssrc;
context->v_base_ts_prev = context->v_last_ts;
context->v_base_ts = timestamp;
context->v_base_seq_prev = context->v_last_seq;
context->v_base_seq = seq;
}
if(context->v_seq_reset) {
/* Video sequence number was paused for a while: just update that */
context->v_seq_reset = FALSE;
context->v_base_seq_prev = context->v_last_seq;
context->v_base_seq = header->seq_number;
}
/* Compute a coherent timestamp and sequence number */
context->v_last_ts = (timestamp-context->v_base_ts) + context->v_base_ts_prev+step;
context->v_last_seq = (seq-context->v_base_seq)+context->v_base_seq_prev+1;
/* Update the timestamp and sequence number in the RTP packet */
header->timestamp = htonl(context->v_last_ts);
header->seq_number = htons(context->v_last_seq);
} else {
if(ssrc != context->a_last_ssrc) {
/* Audio SSRC changed: update both sequence number and timestamp */
JANUS_LOG(LOG_VERB, "Audio SSRC changed, %"SCNu32" --> %"SCNu32"\n",
context->a_last_ssrc, ssrc);
context->a_last_ssrc = ssrc;
context->a_base_ts_prev = context->a_last_ts;
context->a_base_ts = timestamp;
context->a_base_seq_prev = context->a_last_seq;
context->a_base_seq = seq;
}
if(context->a_seq_reset) {
/* Audio sequence number was paused for a while: just update that */
context->a_seq_reset = FALSE;
context->a_base_seq_prev = context->a_last_seq;
context->a_base_seq = header->seq_number;
}
/* Compute a coherent timestamp and sequence number */
context->a_last_ts = (timestamp-context->a_base_ts) + context->a_base_ts_prev+step;
context->a_last_seq = (seq-context->a_base_seq)+context->a_base_seq_prev+1;
/* Update the timestamp and sequence number in the RTP packet */
header->timestamp = htonl(context->a_last_ts);
header->seq_number = htons(context->a_last_seq);
}
}
/* SRTP stuff: we may need our own randomizer */
#ifdef HAVE_SRTP_2
int srtp_crypto_get_random(uint8_t *key, int len) {
/* libsrtp 2.0 doesn't have crypto_get_random, we use OpenSSL's RAND_* to replace it:
* https://wiki.openssl.org/index.php/Random_Numbers */
int rc = RAND_bytes(key, len);
if(rc != 1) {
/* Error generating */
return -1;
}
return 0;
}
#endif
/* SRTP error codes as a string array */
static const char *janus_srtp_error[] =
{
#ifdef HAVE_SRTP_2
"srtp_err_status_ok",
"srtp_err_status_fail",
"srtp_err_status_bad_param",
"srtp_err_status_alloc_fail",
"srtp_err_status_dealloc_fail",
"srtp_err_status_init_fail",
"srtp_err_status_terminus",
"srtp_err_status_auth_fail",
"srtp_err_status_cipher_fail",
"srtp_err_status_replay_fail",
"srtp_err_status_replay_old",
"srtp_err_status_algo_fail",
"srtp_err_status_no_such_op",
"srtp_err_status_no_ctx",
"srtp_err_status_cant_check",
"srtp_err_status_key_expired",
"srtp_err_status_socket_err",
"srtp_err_status_signal_err",
"srtp_err_status_nonce_bad",
"srtp_err_status_read_fail",
"srtp_err_status_write_fail",
"srtp_err_status_parse_err",
"srtp_err_status_encode_err",
"srtp_err_status_semaphore_err",
"srtp_err_status_pfkey_err",
#else
"err_status_ok",
"err_status_fail",
"err_status_bad_param",
"err_status_alloc_fail",
"err_status_dealloc_fail",
"err_status_init_fail",
"err_status_terminus",
"err_status_auth_fail",
"err_status_cipher_fail",
"err_status_replay_fail",
"err_status_replay_old",
"err_status_algo_fail",
"err_status_no_such_op",
"err_status_no_ctx",
"err_status_cant_check",
"err_status_key_expired",
"err_status_socket_err",
"err_status_signal_err",
"err_status_nonce_bad",
"err_status_read_fail",
"err_status_write_fail",
"err_status_parse_err",
"err_status_encode_err",
"err_status_semaphore_err",
"err_status_pfkey_err",
#endif
};
const char *janus_srtp_error_str(int error) {
if(error < 0 || error > 24)
return NULL;
return janus_srtp_error[error];
}