[asterisk-commits] branch oej/test-this-branch r10679 -
/team/oej/test-this-branch/
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Feb 21 13:02:25 MST 2006
Author: oej
Date: Tue Feb 21 14:02:22 2006
New Revision: 10679
URL: http://svn.digium.com/view/asterisk?rev=10679&view=rev
Log:
Adding rtcp patch
Modified:
team/oej/test-this-branch/README.test-this-branch
team/oej/test-this-branch/rtp.c
Modified: team/oej/test-this-branch/README.test-this-branch
URL: http://svn.digium.com/view/asterisk/team/oej/test-this-branch/README.test-this-branch?rev=10679&r1=10678&r2=10679&view=diff
==============================================================================
--- team/oej/test-this-branch/README.test-this-branch (original)
+++ team/oej/test-this-branch/README.test-this-branch Tue Feb 21 14:02:22 2006
@@ -1,7 +1,42 @@
-This branch includes
+Asterisk is developed by the Asterisk.org user community. The
+development team does not only consist of coders, but also
+testers and people that write documentation and check for
+security problems.
+
+This is a combined branch of many patches and branches from the
+bug tracker that needs your testing. Please test and report
+your results in the bug tracker reports for each patch.
+
+This branch includes the following branches
- sipdiversion: Additional support for the Diversion: header
-- jitterbuffer: Jitterbuffer for RTP in chan_sip
-- New CLI commands for global variables
+- jitterbuffer: Jitterbuffer for RTP in chan_sip (#3854)
+- videosupport: Improved support for video (#5427)
+- peermatch: New peer matching algorithm (no bug report yet)
+- rtcp: Improved support for RTCP (#2863)
+
+Coming here soon:
+- sipregister: A new registration architecture (#5834)
+- iptos: New IPtos support, separate audio and signalling (#6355)
+- metermaids: Subscription support for parking lots (#5779)
+
+And the following stand-alone patches
+- New CLI commands for global variables (#6506)
- Additional options for the CHANNEL dialplan function
-- videosupport: Improved support for video
+- Report dialplan reload in manager (#5741)
+
+All of these exist in the bug tracker
+
+* PEERMATCH: New object match for incoming calls. Skip the "user" :-)
+---------------------------------------------------------------------
+In this code, we will match incoming calls like this:
+
+- First user on From: user name
+- Then peer on From: user name *** NEW ****
+- Then peer on IP and port
+
+
+This means that in most configurations, you can configure a phone entry as
+"type=peer" instead of "type=friend". Subscriptions will work much better
+with just one object to match.
+
-
Modified: team/oej/test-this-branch/rtp.c
URL: http://svn.digium.com/view/asterisk/team/oej/test-this-branch/rtp.c?rev=10679&r1=10678&r2=10679&view=diff
==============================================================================
--- team/oej/test-this-branch/rtp.c (original)
+++ team/oej/test-this-branch/rtp.c Tue Feb 21 14:02:22 2006
@@ -58,6 +58,16 @@
#include "asterisk/utils.h"
#define MAX_TIMESTAMP_SKEW 640
+#define RTP_SEQ_MOD (1<<16) /*A sequence number can't be more than 16 bits */
+#define RTCP_DEFAULT_INTERVALMS 5000 /* Default milli-seconds between RTCP reports we send */
+#define RTCP_MIN_INTERVALMS 500 /* Min milli-seconds between RTCP reports we send */
+#define RTCP_MAX_INTERVALMS 60000 /* Max milli-seconds between RTCP reports we send */
+#define RTCP_PT_FUR 192
+#define RTCP_PT_SR 200
+#define RTCP_PT_RR 201
+#define RTCP_PT_SDES 202
+#define RTCP_PT_BYE 203
+#define RTCP_PT_APP 204
#define RTP_MTU 1200
@@ -68,10 +78,17 @@
static int rtpstart = 0; /*!< First port for RTP sessions (set in rtp.conf) */
static int rtpend = 0; /*!< Last port for RTP sessions (set in rtp.conf) */
static int rtpdebug = 0; /*!< Are we debugging? */
+static int rtcpdebug = 0; /*!< Are we debugging RTCP? */
+static int rtcpstats = 0; /*!< Are we debugging RTCP? */
+static int rtcpinterval = RTCP_DEFAULT_INTERVALMS; /*!< Time between rtcp reports in millisecs */
static struct sockaddr_in rtpdebugaddr; /*!< Debug packets to/from this host */
+static struct sockaddr_in rtcpdebugaddr; /*!< Debug RTCP packets to/from this host */
#ifdef SO_NO_CHECK
static int nochecksums = 0;
#endif
+
+/* Forward declarations */
+int ast_rtcp_write(void *data);
/*! \brief The value of each payload format mapping: */
struct rtpPayloadType {
@@ -93,12 +110,23 @@
struct ast_frame f;
unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET];
unsigned int ssrc; /*!< Synchronization source, RFC 3550, page 10. */
+ unsigned int themssrc; /* Their SSRC */
unsigned int lastts;
unsigned int lastdigitts;
unsigned int lastrxts;
unsigned int lastividtimestamp;
unsigned int lastovidtimestamp;
unsigned int lasteventseqn;
+ int lastrxseqno; /* Last received sequence number */
+ unsigned short seedrxseqno; /* What sequence number did they start with?*/
+ unsigned int seedrxts; /* What RTP timestamp did they start with? */
+ unsigned int rxcount; /* How many packets have we received? */
+ unsigned int rxoctetcount; /* How many octets have we received? should be rxcount *160*/
+ unsigned int txcount; /* How many packets have we sent? */
+ unsigned int txoctetcount; /* How many octets have we sent? (txcount*160)*/
+ unsigned int cycles; /* Shifted count of sequence number cycles */
+ double rxjitter; /* Interarrival jitter at the moment */
+ double rxtransit; /* Relative transit time for previous packet */
unsigned int lasteventendseqn;
int lasttxformat;
int lastrxformat;
@@ -110,6 +138,8 @@
struct sockaddr_in them; /*!< Socket representation of the remote endpoint. */
struct timeval rxcore;
struct timeval txcore;
+ double drxcore; /* The double representation of the first received packet */
+ struct timeval lastrx; /* timeval when we last received a packet */
struct timeval dtmfmute;
struct ast_smoother *smoother;
int *ioid;
@@ -123,6 +153,7 @@
int rtp_lookup_code_cache_isAstFormat; /*!< a cache for the result of rtp_lookup_code(): */
int rtp_lookup_code_cache_code;
int rtp_lookup_code_cache_result;
+ int rtp_offered_from_local;
struct ast_rtcp *rtcp;
};
@@ -140,10 +171,41 @@
int s; /*!< Socket */
struct sockaddr_in us; /*!< Socket representation of the local endpoint. */
struct sockaddr_in them; /*!< Socket representation of the remote endpoint. */
+ unsigned int soc; /* What they told us */
+ unsigned int spc; /* What they told us */
+ unsigned int themrxlsr; /* The middle 32 bits of the NTP timestamp in the last received SR*/
+ struct timeval rxlsr; /* Time when we got their last SR */
+ struct timeval txlsr; /* Time when we sent or last SR*/
+ unsigned int expected_prior; /* no. packets in previous interval */
+ unsigned int received_prior; /* no. packets received in previous interval */
+ int schedid; /*Schedid returned from ast_sched_add() to schedule RTCP-transmissions*/
+ unsigned int rr_count; /* number of RRs we've sent, not including report blocks in SR's */
+ unsigned int sr_count; /* number of SRs we've sent */
+ unsigned int lastsrtxcount; /* Transmit packet count when last SR sent */
+ double accumulated_transit; /* accumulated a-dlsr-lsr */
+ double rtt; /* Last reported rtt */
+ unsigned int reported_jitter; /* The contents of their last jitter entry in the RR */
+ unsigned int reported_lost; /* Reported lost packets in their RR */
+ char quality[AST_MAX_USER_FIELD];
+ double maxrxjitter;
+ double minrxjitter;
+ double maxrtt;
+ double minrtt;
+ int sendfur;
};
/*! \brief List of current sessions */
static AST_LIST_HEAD_STATIC(protos, ast_rtp_protocol);
+
+void timeval2ntp(struct timeval tv, unsigned int *msw, unsigned int *lsw)
+{
+ unsigned int sec, usec, frac;
+ sec = tv.tv_sec + 2208988800u; /* Sec between 1900 and 1970 */
+ usec = tv.tv_usec;
+ frac = (usec << 12) + (usec << 8) - ((usec * 3650) >> 6);
+ *msw = sec;
+ *lsw = frac;
+}
int ast_rtp_fd(struct ast_rtp *rtp)
{
@@ -155,6 +217,15 @@
if (rtp->rtcp)
return rtp->rtcp->s;
return -1;
+}
+
+unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp)
+{
+ unsigned int interval;
+ /* TODO Do a more reasonable calculation on this one
+ * Look in RFC 3550 Section A.7 for an example*/
+ interval = rtcpinterval;
+ return interval;
}
void ast_rtp_set_data(struct ast_rtp *rtp, void *data)
@@ -214,6 +285,20 @@
}
return 1;
}
+
+static inline int rtcp_debug_test_addr(struct sockaddr_in *addr)
+{
+ if (rtcpdebug == 0)
+ return 0;
+ if (rtcpdebugaddr.sin_addr.s_addr) {
+ if (((ntohs(rtcpdebugaddr.sin_port) != 0)
+ && (rtcpdebugaddr.sin_port != addr->sin_port))
+ || (rtcpdebugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
+ return 0;
+ }
+ return 1;
+}
+
static struct ast_frame *process_cisco_dtmf(struct ast_rtp *rtp, unsigned char *data, int len)
{
@@ -364,33 +449,43 @@
struct ast_frame *ast_rtcp_read(struct ast_rtp *rtp)
{
socklen_t len;
- int hdrlen = 8;
+ int position, i, packetwords;
int res;
struct sockaddr_in sin;
- unsigned int rtcpdata[1024];
+ unsigned int rtcpdata[8192 + AST_FRIENDLY_OFFSET];
char iabuf[INET_ADDRSTRLEN];
+ unsigned int *rtcpheader;
+ int pt;
+ struct timeval now;
+ unsigned int length;
+ int rc;
+ double rtt = 0;
+ double a;
+ double dlsr;
+ double lsr;
+ unsigned int msw;
+ unsigned int lsw;
+ unsigned int comp;
if (!rtp || !rtp->rtcp)
return &ast_null_frame;
len = sizeof(sin);
- res = recvfrom(rtp->rtcp->s, rtcpdata, sizeof(rtcpdata),
+ res = recvfrom(rtp->rtcp->s, rtcpdata + AST_FRIENDLY_OFFSET, sizeof(rtcpdata) - AST_FRIENDLY_OFFSET,
0, (struct sockaddr *)&sin, &len);
+ rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET);
if (res < 0) {
if (errno != EAGAIN)
- ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno));
+ ast_log(LOG_WARNING, "RTCP Read error: %s\n", strerror(errno));
if (errno == EBADF)
CRASH;
return &ast_null_frame;
}
- if (res < hdrlen) {
- ast_log(LOG_WARNING, "RTP Read too short\n");
- return &ast_null_frame;
- }
-
+ packetwords = res / 4;
+
if (rtp->nat) {
/* Send to whoever sent to us */
if ((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
@@ -402,18 +497,147 @@
}
if (option_debug)
ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res);
+
+ /* Process a compound packet */
+ position = 0;
+ while (position < packetwords) {
+ i = position;
+ length = ntohl(rtcpheader[i]);
+ pt = (length & 0xff0000) >> 16;
+ rc = (length & 0x1f000000) >> 24;
+ length &= 0xffff;
+
+ if ((i + length) > packetwords) {
+ ast_log(LOG_WARNING, "RTCP Read too short\n");
+ return &ast_null_frame;
+ }
+
+ if(rtcp_debug_test_addr(&sin)){
+ ast_verbose("\n\nGot RTCP from %s:%d\n",ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr),ntohs(sin.sin_port));
+ ast_verbose("PT: %d(%s)\n",pt,(pt==200)?"Sender Report":(pt==201)?"Receiver Report":(pt==192)?"H.261 FUR":"Unknown");
+ ast_verbose("Reception reports: %d\n",rc);
+ ast_verbose("SSRC of sender: %u\n",rtcpheader[i+1]);
+ }
+
+ i+=2; /* Advance past header and ssrc */
+
+ switch(pt){
+ case RTCP_PT_SR:
+ gettimeofday(&rtp->rtcp->rxlsr,NULL); /* To be able to populate the dlsr */
+ rtp->rtcp->spc = ntohl(rtcpheader[i+3]);
+ rtp->rtcp->soc = ntohl(rtcpheader[i+4]);
+ rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i+1]) & 0xffff) >> 16); /* Going to LSR in RR*/
+
+ if(rtcp_debug_test_addr(&sin)){
+ ast_verbose("NTP timestamp: %u.%010u\n",ntohl(rtcpheader[i]), ntohl(rtcpheader[i+1])*4096);
+ ast_verbose("RTP timestamp: %u\n",ntohl(rtcpheader[i+2]));
+ ast_verbose("SPC: %u\tSOC: %u\n",ntohl(rtcpheader[i+3]),ntohl(rtcpheader[i+4]));
+ }
+ i += 5;
+ if (rc < 1)
+ break;
+ /* Intentional fall through */
+ case RTCP_PT_RR:
+ /* This is the place to calculate RTT */
+ /* Don't handle multiple reception reports (rc > 1) yet */
+ gettimeofday(&now, NULL);
+ timeval2ntp(now, &msw, &lsw);
+ /* Use the one we sent them in our SR instead, rtcp->txlsr could have been rewritten if the dlsr is large */
+ if(ntohl(rtcpheader[i+4])){ /* We must have the LSR */
+ comp = ((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16);
+ a = (double)((comp & 0xffff0000) >> 16) + (double)((double)(comp & 0xffff)/1000000.);
+ lsr = (double)((ntohl(rtcpheader[i+4]) & 0xffff0000) >> 16) + (double)((double)(ntohl(rtcpheader[i+4]) & 0xffff)/1000000.);
+ dlsr = (double)(ntohl(rtcpheader[i+5])/65536.);
+ rtt = a - dlsr - lsr;
+ rtp->rtcp->accumulated_transit += rtt;
+ rtp->rtcp->rtt = rtt;
+ if(rtp->rtcp->maxrtt<rtt)
+ rtp->rtcp->maxrtt = rtt;
+ if(rtp->rtcp->minrtt>rtt)
+ rtp->rtcp->minrtt = rtt;
+ }
+ rtp->rtcp->reported_jitter = ntohl(rtcpheader[i+3]);
+ rtp->rtcp->reported_lost = ntohl(rtcpheader[i+1]) & 0xffffff;
+ if(rtcp_debug_test_addr(&sin)){
+ ast_verbose("Fraction lost: %d\n", ((ntohl(rtcpheader[i+1]) & 0xff000000) >> 24));
+ ast_verbose("Packets lost so far: %d\n", rtp->rtcp->reported_lost);
+ ast_verbose("Highest sequence number: %d\n", (ntohl(rtcpheader[i+2]) & 0xffff));
+ ast_verbose("Sequence number cycles: %d\n", (ntohl(rtcpheader[i+2]) & 0xffff) >> 16);
+ ast_verbose("Interarrival jitter: %u\n", rtp->rtcp->reported_jitter);
+ ast_verbose("Last SR(our NTP): %u.%010u\n",ntohl(rtcpheader[i+4])>>16,(ntohl(rtcpheader[i+4])<<16)*4096);
+ ast_verbose("DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i+5])/65536.0);
+ if(rtt)
+ ast_verbose("RTT: %f(sec)\n", rtt);
+ }
+ break;
+ case RTCP_PT_FUR:
+ if(rtcp_debug_test_addr(&sin))
+ ast_verbose("Received a Fast Update Request\n");
+ break;
+ case RTCP_PT_SDES:
+ if(rtcp_debug_test_addr(&sin))
+ ast_verbose("Received an SDES from %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
+ break;
+ case RTCP_PT_BYE:
+ if(rtcp_debug_test_addr(&sin))
+ ast_verbose("Received a BYE from %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
+ break;
+ default:
+ ast_log(LOG_NOTICE, "Unknown RTCP packet (pt=%d) received from %s:%d\n", pt, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
+ break;
+ }
+ position += (length + 1);
+ }
+
return &ast_null_frame;
}
static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int timestamp, int mark)
{
- struct timeval ts = ast_samp2tv( timestamp, 8000);
- if (ast_tvzero(rtp->rxcore) || mark) {
- rtp->rxcore = ast_tvsub(ast_tvnow(), ts);
- /* Round to 20ms for nice, pretty timestamps */
- rtp->rxcore.tv_usec -= rtp->rxcore.tv_usec % 20000;
- }
- *tv = ast_tvadd(rtp->rxcore, ts);
+ struct timeval now;
+ double transit;
+ double current_time;
+ double d;
+ double dtv;
+ double prog;
+
+ if ((!rtp->rxcore.tv_sec && !rtp->rxcore.tv_usec) || mark) {
+ gettimeofday(&rtp->rxcore, NULL);
+ rtp->drxcore = (double)rtp->rxcore.tv_sec + (double)rtp->rxcore.tv_usec/1000000;
+ /* map timestamp to a real time */
+ rtp->seedrxts = timestamp; /* Their RTP timestamp started with this */
+ rtp->rxcore.tv_sec -= timestamp / 8000;
+ rtp->rxcore.tv_usec -= (timestamp % 8000) * 125;
+ /* Round to 0.1ms for nice, pretty timestamps */
+ rtp->rxcore.tv_usec -= rtp->rxcore.tv_usec % 100;
+ if (rtp->rxcore.tv_usec < 0) {
+ /* Adjust appropriately if necessary */
+ rtp->rxcore.tv_usec += 1000000;
+ rtp->rxcore.tv_sec -= 1;
+ }
+ }
+
+ gettimeofday(&now,NULL);
+ /* rxcore is the mapping between the RTP timestamp and _our_ real time from gettimeofday() */
+ tv->tv_sec = rtp->rxcore.tv_sec + timestamp / 8000;
+ tv->tv_usec = rtp->rxcore.tv_usec + (timestamp % 8000) * 125;
+ if (tv->tv_usec >= 1000000) {
+ tv->tv_usec -= 1000000;
+ tv->tv_sec += 1;
+ }
+ prog = (double)((timestamp-rtp->seedrxts)/8000.);
+ dtv = (double)rtp->drxcore + (double)(prog);
+ current_time = (double)now.tv_sec + (double)now.tv_usec/1000000;
+ transit = current_time - dtv;
+ d = transit - rtp->rxtransit;
+ rtp->rxtransit = transit;
+ if(d<0)
+ d=-d;
+ rtp->rxjitter += (1./16.) * (d - rtp->rxjitter);
+ if(rtp->rxjitter > rtp->rtcp->maxrxjitter)
+ rtp->rtcp->maxrxjitter = rtp->rxjitter;
+ if(rtp->rxjitter < rtp->rtcp->minrxjitter)
+ rtp->rtcp->minrxjitter = rtp->rxjitter;
}
struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
@@ -424,6 +648,7 @@
unsigned int seqno;
int version;
int payloadtype;
+ int tseqno;
int hdrlen = 12;
int padding;
int mark;
@@ -453,6 +678,7 @@
CRASH;
return &ast_null_frame;
}
+
if (res < hdrlen) {
ast_log(LOG_WARNING, "RTP Read too short\n");
return &ast_null_frame;
@@ -468,6 +694,10 @@
if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
(rtp->them.sin_port != sin.sin_port)) {
memcpy(&rtp->them, &sin, sizeof(rtp->them));
+ if(rtp->rtcp){
+ memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
+ rtp->rtcp->them.sin_port = htons(ntohs(rtp->them.sin_port)+1);
+ }
rtp->rxseqno = 0;
ast_set_flag(rtp, FLAG_NAT_ACTIVE);
if (option_debug || rtpdebug)
@@ -506,11 +736,35 @@
return &ast_null_frame;
}
+ rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */
+
+ tseqno = rtp->lastrxseqno +1;
+
+ if(rtp->rxcount==1){
+ /* This is the first RTP packet successfully received from source */
+ rtp->seedrxseqno = seqno;
+ }
+
+ if(rtp->rtcp->schedid<1){
+ /* Schedule transmission of Receiver Report */
+ rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
+ }
+
+ if(tseqno > RTP_SEQ_MOD){ /* if tseqno is greater than RTP_SEQ_MOD it would indicate that the sender cycled */
+ rtp->cycles += RTP_SEQ_MOD;
+ ast_verbose("SEQNO cycled: %u\t%d\n", rtp->cycles, seqno);
+ }
+
+ rtp->lastrxseqno = seqno;
+
+ if(rtp->themssrc==0)
+ rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
+
if(rtp_debug_test_addr(&sin))
ast_verbose("Got RTP packet from %s:%d (type %d, seq %d, ts %d, len %d)\n"
, ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
- rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
+ rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
if (!rtpPT.isAstFormat) {
/* This is special in-band data that's not one of our codecs */
if (rtpPT.code == AST_RTP_DTMF) {
@@ -872,6 +1126,13 @@
}
}
+void ast_rtp_offered_from_local(struct ast_rtp* rtp, int local) {
+ if (rtp)
+ rtp->rtp_offered_from_local = local;
+ else
+ ast_log(LOG_WARNING, "rtp structure is null\n");
+}
+
struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt)
{
struct rtpPayloadType result;
@@ -881,7 +1142,8 @@
return result; /* bogus payload type */
/* Start with the negotiated codecs */
- result = rtp->current_RTP_PT[pt];
+ if (!rtp->rtp_offered_from_local)
+ result = rtp->current_RTP_PT[pt];
/* If it doesn't exist, check our static RTP type list, just in case */
if (!result.code)
@@ -1001,9 +1263,11 @@
memset(rtcp, 0, sizeof(struct ast_rtcp));
rtcp->s = rtp_socket();
rtcp->us.sin_family = AF_INET;
+ rtcp->them.sin_family = AF_INET;
+
if (rtcp->s < 0) {
free(rtcp);
- ast_log(LOG_WARNING, "Unable to allocate socket: %s\n", strerror(errno));
+ ast_log(LOG_WARNING, "Unable to allocate RTCP socket: %s\n", strerror(errno));
return NULL;
}
return rtcp;
@@ -1138,11 +1402,16 @@
void ast_rtp_stop(struct ast_rtp *rtp)
{
+ if(rtp->rtcp->schedid>0){
+ ast_sched_del(rtp->sched, rtp->rtcp->schedid);
+ rtp->rtcp->schedid = -1;
+ }
+
memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port));
if (rtp->rtcp) {
- memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
- memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->them.sin_port));
+ memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr));
+ memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port));
}
}
@@ -1166,8 +1435,50 @@
rtp->rxseqno = 0;
}
+char *ast_rtp_get_quality(struct ast_rtp *rtp)
+{
+ /*
+ *ssrc our ssrc
+ *themssrc their ssrc
+ *lp lost packets
+ *rxjitter our calculated jitter(rx)
+ *rxcount no. received packets
+ *txjitter reported jitter of the other end
+ *txcount transmitted packets
+ *rlp remote lost packets
+ */
+
+ snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality), "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f", rtp->ssrc, rtp->themssrc, rtp->rtcp->expected_prior - rtp->rtcp->received_prior, rtp->rxjitter, rtp->rxcount, (double)rtp->rtcp->reported_jitter/65536., rtp->txcount, rtp->rtcp->reported_lost, rtp->rtcp->rtt);
+
+ return rtp->rtcp->quality;
+}
+
void ast_rtp_destroy(struct ast_rtp *rtp)
{
+ if(rtcp_debug_test_addr(&rtp->them) || rtcpstats){
+ /*Print some info on the call here */
+ ast_verbose(" RTP-stats\n");
+ ast_verbose("* Our Receiver:\n");
+ ast_verbose(" SSRC: %u\n", rtp->themssrc);
+ ast_verbose(" Received packets: %u\n", rtp->rxcount);
+ ast_verbose(" Lost packets: %u\n", rtp->rtcp->expected_prior - rtp->rtcp->received_prior);
+ ast_verbose(" Jitter: %.4f\n", rtp->rxjitter);
+ ast_verbose(" Transit: %.4f\n", rtp->rxtransit);
+ ast_verbose(" RR-count: %u\n", rtp->rtcp->rr_count);
+ ast_verbose("* Our Sender:\n");
+ ast_verbose(" SSRC: %u\n", rtp->ssrc);
+ ast_verbose(" Sent packets: %u\n", rtp->txcount);
+ ast_verbose(" Lost packets: %u\n", rtp->rtcp->reported_lost);
+ ast_verbose(" Jitter: %u\n", rtp->rtcp->reported_jitter);
+ ast_verbose(" SR-count: %u\n", rtp->rtcp->sr_count);
+ ast_verbose(" RTT: %f\n", rtp->rtcp->rtt);
+ }
+
+ if(rtp->rtcp->schedid>0){
+ ast_sched_del(rtp->sched, rtp->rtcp->schedid);
+ rtp->rtcp->schedid = -1;
+ }
+
if (rtp->smoother)
ast_smoother_free(rtp->smoother);
if (rtp->ioid)
@@ -1177,6 +1488,7 @@
if (rtp->rtcp) {
close(rtp->rtcp->s);
free(rtp->rtcp);
+ rtp->rtcp=NULL;
}
free(rtp);
}
@@ -1282,6 +1594,238 @@
return 0;
}
+/* Send an H.261 fast update request, some devices need this rather than SIP XML */
+int ast_rtcp_send_h261fur(void *data)
+{
+ struct ast_rtp *rtp = data;
+ int res;
+
+ rtp->rtcp->sendfur = 1;
+ res = ast_rtcp_write(data);
+
+ return res;
+}
+
+int ast_rtcp_write_sr(void *data)
+{
+ struct ast_rtp *rtp = data;
+ int res;
+ int len = 0;
+ struct timeval now;
+ unsigned int now_lsw;
+ unsigned int now_msw;
+ unsigned int *rtcpheader;
+ unsigned int lost;
+ unsigned int extended;
+ unsigned int expected;
+ unsigned int expected_interval;
+ unsigned int received_interval;
+ unsigned int lost_interval;
+ int fraction;
+ struct timeval dlsr;
+ char bdata[512];
+ char iabuf[INET_ADDRSTRLEN];
+
+ if(!rtp || !rtp->rtcp || (&rtp->rtcp->them.sin_addr == 0))
+ return 0;
+
+ if(!rtp->rtcp->them.sin_addr.s_addr){ /* This'll stop rtcp for this rtp session */
+ ast_verbose("RTCP SR transmission error, rtcp halted %s\n",strerror(errno));
+ ast_sched_del(rtp->sched, rtp->rtcp->schedid);
+ rtp->rtcp->schedid = -1;
+ return 0;
+ }
+
+ gettimeofday(&now, NULL);
+ timeval2ntp(now, &now_msw, &now_lsw); /* fill thses ones in from utils.c*/
+ rtcpheader = (unsigned int *)bdata;
+ rtcpheader[1] = htonl(rtp->ssrc); /* Our SSRC */
+ rtcpheader[2] = htonl(now_msw); /* now, MSW. gettimeofday() + SEC_BETWEEN_1900_AND_1970*/
+ rtcpheader[3] = htonl(now_lsw); /* now, LSW */
+ rtcpheader[4] = htonl(rtp->lastts); /* FIXME shouldn't be that, it should be now */
+ rtcpheader[5] = htonl(rtp->txcount); /* No. packets sent */
+ rtcpheader[6] = htonl(rtp->txoctetcount); /* No. bytes sent */
+ len += 28;
+
+ extended = rtp->cycles + rtp->lastrxseqno;
+ expected = extended - rtp->seedrxseqno + 1;
+ if(rtp->rxcount > expected) {
+ expected += rtp->rxcount - expected;
+ }
+ lost = expected - rtp->rxcount;
+ expected_interval = expected - rtp->rtcp->expected_prior;
+ rtp->rtcp->expected_prior = expected;
+ received_interval = rtp->rxcount - rtp->rtcp->received_prior;
+ rtp->rtcp->received_prior = rtp->rxcount;
+ lost_interval = expected_interval - received_interval;
+ if(expected_interval == 0 || lost_interval <= 0)
+ fraction = 0;
+ else
+ fraction = (lost_interval << 8) / expected_interval;
+ timersub(&now, &rtp->rtcp->rxlsr, &dlsr);
+ rtcpheader[7] = htonl(rtp->themssrc);
+ rtcpheader[8] = htonl(((fraction & 0xff) << 24) | (lost & 0xffffff));
+ rtcpheader[9] = htonl((rtp->cycles << 16) | ((rtp->lastrxseqno & 0xffff)));
+ rtcpheader[10] = htonl((unsigned int)rtp->rxjitter);
+ rtcpheader[11] = htonl(rtp->rtcp->themrxlsr);
+ rtcpheader[12] = htonl((((dlsr.tv_sec * 1000) + (dlsr.tv_usec / 1000)) * 65536) / 1000);
+ len += 24;
+
+ rtcpheader[0] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SR << 16) | ((len/4)-1));
+
+ if (rtp->rtcp->sendfur) {
+ rtcpheader[13] = htonl((2 << 30) | (0 << 24) | (RTCP_PT_FUR << 16) | 1);
+ rtcpheader[14] = htonl(rtp->ssrc); /* Our SSRC */
+ len += 8;
+ rtp->rtcp->sendfur = 0;
+ }
+
+ /* Insert SDES here. Probably should make SDES text equal to mimetypes[code].type (not subtype 'cos */
+ /* it can change mid call, and SDES can't) */
+ rtcpheader[len/4] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SDES << 16) | 2);
+ rtcpheader[(len/4)+1] = htonl(rtp->ssrc); /* Our SSRC */
+ rtcpheader[(len/4)+2] = htonl(0x01 << 24); /* Empty for the moment */
+ len += 12;
+
+ res = sendto(rtp->rtcp->s, (unsigned int *)rtcpheader, len, 0, (struct sockaddr *)&rtp->rtcp->them, sizeof(rtp->rtcp->them));
+ if(res<0){
+ ast_verbose("RTCP SR transmission error to %s:%d, rtcp halted %s\n",ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port), strerror(errno));
+ ast_sched_del(rtp->sched, rtp->rtcp->schedid);
+ rtp->rtcp->schedid = -1;
+ return 0;
+ }
+
+ /* FIXME Don't need to get a new one */
+ gettimeofday(&rtp->rtcp->txlsr, NULL);
+ rtp->rtcp->sr_count++;
+
+ rtp->rtcp->lastsrtxcount = rtp->txcount;
+
+ if(rtcp_debug_test_addr(&rtp->rtcp->them)){
+ ast_verbose("Sent RTCP SR to %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
+ ast_verbose("Our SSRC: %u\n", rtp->ssrc);
+ ast_verbose("Sent(NTP): %u.%010u\n", (unsigned int)now.tv_sec, (unsigned int)now.tv_usec*4096);
+ ast_verbose("Sent(RTP): %u\n", rtp->lastts);
+ ast_verbose("Sent packets: %u\n", rtp->txcount);
+ ast_verbose("Sent octets: %u\n", rtp->txoctetcount);
+ ast_verbose("Report block:\n");
+ ast_verbose("Fraction lost: %u\n", fraction);
+ ast_verbose("Cumulative loss: %u\n", lost);
+ ast_verbose("IA jitter: %.4f\n", rtp->rxjitter);
+ ast_verbose("Their last SR: %u\n", rtp->rtcp->themrxlsr);
+ ast_verbose("DLSR: %4.4f (sec)\n\n", (double)(ntohl(rtcpheader[12])/65536.0));
+ }
+ return res;
+}
+
+int ast_rtcp_write_rr(void *data)
+{
+ struct ast_rtp *rtp = data;
+ int res;
+ int len = 32;
+ unsigned int lost;
+ unsigned int extended;
+ unsigned int expected;
+ unsigned int expected_interval;
+ unsigned int received_interval;
+ unsigned int lost_interval;
+ struct timeval now;
+ unsigned int *rtcpheader;
+ char bdata[1024];
+ char iabuf[INET_ADDRSTRLEN];
+ struct timeval dlsr;
+ int fraction;
+
+ if(!rtp || !rtp->rtcp || (&rtp->rtcp->them.sin_addr == 0))
+ return 0;
+
+ if(!rtp->rtcp->them.sin_addr.s_addr){
+ ast_verbose("RTCP RR transmission error to, rtcp halted %s\n",strerror(errno));
+ ast_sched_del(rtp->sched, rtp->rtcp->schedid);
+ rtp->rtcp->schedid = -1;
+ return 0;
+ }
+
+ extended = rtp->cycles + rtp->lastrxseqno;
+ expected = extended - rtp->seedrxseqno + 1;
+ lost = expected - rtp->rxcount;
+ expected_interval = expected - rtp->rtcp->expected_prior;
+ rtp->rtcp->expected_prior = expected;
+ received_interval = rtp->rxcount - rtp->rtcp->received_prior;
+ rtp->rtcp->received_prior = rtp->rxcount;
+ lost_interval = expected_interval - received_interval;
+ if(expected_interval == 0 || lost_interval <= 0)
+ fraction = 0;
+ else
+ fraction = (lost_interval << 8) / expected_interval;
+ gettimeofday(&now, NULL);
+ timersub(&now, &rtp->rtcp->rxlsr, &dlsr);
+ rtcpheader = (unsigned int *)bdata;
+ rtcpheader[0] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_RR << 16) | ((len/4)-1));
+ rtcpheader[1] = htonl(rtp->ssrc);
+ rtcpheader[2] = htonl(rtp->themssrc);
+ rtcpheader[3] = htonl(((fraction & 0xff) << 24) | (lost & 0xffffff));
+ rtcpheader[4] = htonl((rtp->cycles << 16) | ((rtp->lastrxseqno & 0xffff)));
+ rtcpheader[5] = htonl((unsigned int)rtp->rxjitter);
+ rtcpheader[6] = htonl(rtp->rtcp->themrxlsr);
+ rtcpheader[7] = htonl((((dlsr.tv_sec * 1000) + (dlsr.tv_usec / 1000)) * 65536) / 1000);
+
+ if (rtp->rtcp->sendfur) {
+ rtcpheader[8] = htonl((2 << 30) | (0 << 24) | (RTCP_PT_FUR << 16) | 1); /* Header from page 36 in RFC 3550 */
+ rtcpheader[9] = htonl(rtp->ssrc); /* Our SSRC */
+ len += 8;
+ rtp->rtcp->sendfur = 0;
+ }
+
+ /* Insert SDES here. Probably should make SDES text equal to mimetypes[code].type (not subtype 'cos */
+ /* it can change mid call, and SDES can't) */
+ rtcpheader[len/4] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SDES << 16) | 2);
+ rtcpheader[(len/4)+1] = htonl(rtp->ssrc); /* Our SSRC */
+ rtcpheader[(len/4)+2] = htonl(0x01 << 24); /* Empty for the moment */
+ len += 12;
+
+ res = sendto(rtp->rtcp->s, (unsigned int *)rtcpheader, len, 0, (struct sockaddr *)&rtp->rtcp->them, sizeof(rtp->rtcp->them));
+
+ if(res<0){
+ ast_verbose("RTCP RR transmission error, rtcp halted: %s\n",strerror(errno));
+ /* Remove the scheduler */
+ ast_sched_del(rtp->sched, rtp->rtcp->schedid);
+ rtp->rtcp->schedid = -1;
+ return 0;
+ }
+
+ rtp->rtcp->rr_count++;
+
+ if(rtcp_debug_test_addr(&rtp->rtcp->them)){
+ ast_verbose("\nSending RTCP RR to %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
+ ast_verbose("Our SSRC: %u\n", rtp->ssrc);
+ ast_verbose("Their SSRC: %u\n", rtp->themssrc);
+ ast_verbose("Fraction lost: %d\n", fraction);
+ ast_verbose("Cumulative loss: %u\n", lost);
+ ast_verbose("IA jitter: %.4f\n", rtp->rxjitter);
+ ast_verbose("Their last SR: %u\n", rtp->rtcp->themrxlsr);
+ ast_verbose("DLSR: %4.4f (sec)\n\n", (double)(ntohl(rtcpheader[7])/65536.0));
+ }
+
+ return res;
+}
+
+/* Write and RTCP packet to the far end */
+/* Decide if we are going to send an SR (with Reception Block) or RR */
+/* RR is sent if we have not sent any rtp packets in the previous interval */
+int ast_rtcp_write(void *data)
+{
+ struct ast_rtp *rtp = data;
+ int res;
+
+ if (rtp->txcount > rtp->rtcp->lastsrtxcount)
+ res = ast_rtcp_write_sr(data);
+ else
+ res = ast_rtcp_write_rr(data);
+
+ return res;
+}
+
int ast_rtp_sendcng(struct ast_rtp *rtp, int level)
{
unsigned int *rtpheader;
@@ -1310,7 +1854,7 @@
if (res <0)
ast_log(LOG_ERROR, "RTP Comfort Noise Transmission error to %s:%d: %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
if(rtp_debug_test_addr(&rtp->them))
- ast_verbose("Sent Comfort Noise RTP packet to %s:%d (type %d, seq %d, ts %d, len %d)\n"
+ ast_verbose("Sent Comfort Noise RTP packet to %s:%d (type %d, seq %d, ts %u, len %d)\n"
, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen);
}
@@ -1385,6 +1929,13 @@
if (option_debug || rtpdebug)
ast_log(LOG_DEBUG, "RTP NAT: Can't write RTP to private address %s:%d, waiting for other end to send audio...\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port));
ast_set_flag(rtp, FLAG_NAT_INACTIVE_NOWARN);
+ }
+ }else{
+ rtp->txcount++;
+ rtp->txoctetcount +=(res - hdrlen);
+
+ if(rtp->rtcp->schedid<1){
+ rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
}
}
@@ -1845,6 +2396,37 @@
return RESULT_SUCCESS;
}
+static int rtcp_do_debug_ip(int fd, int argc, char *argv[])
+{
+ struct hostent *hp;
+ struct ast_hostent ahp;
+ char iabuf[INET_ADDRSTRLEN];
+ int port = 0;
+ char *p, *arg;
+ if (argc != 5)
+ return RESULT_SHOWUSAGE;
+
+ arg = argv[4];
+ p = strstr(arg, ":");
+ if (p){
+ *p = '\0';
+ p++;
+ port = atoi(p);
+ }
+ hp = ast_gethostbyname(arg, &ahp);
+ if (hp == NULL)
+ return RESULT_SHOWUSAGE;
+ rtcpdebugaddr.sin_family = AF_INET;
+ memcpy(&rtcpdebugaddr.sin_addr, hp->h_addr, sizeof(rtcpdebugaddr.sin_addr));
+ rtcpdebugaddr.sin_port = htons(port);
+ if (port == 0)
+ ast_cli(fd, "RTCP Debugging Enabled for IP: %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtcpdebugaddr.sin_addr));
+ else
+ ast_cli(fd, "RTCP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtcpdebugaddr.sin_addr), port);
+ rtcpdebug = 1;
+ return RESULT_SUCCESS;
+}
+
static int rtp_do_debug(int fd, int argc, char *argv[])
{
if(argc != 2) {
@@ -1858,6 +2440,27 @@
return RESULT_SUCCESS;
}
+static int rtcp_do_debug(int fd, int argc, char *argv[]){
+ if(argc != 3){
+ if(argc != 5)
+ return RESULT_SHOWUSAGE;
+ return rtcp_do_debug_ip(fd, argc, argv);
+ }
+ rtcpdebug = 1;
+ memset(&rtcpdebugaddr,0,sizeof(rtcpdebugaddr));
+ ast_cli(fd, "RTCP Debugging Enabled\n");
+ return RESULT_SUCCESS;
+}
+
+static int rtcp_do_stats(int fd, int argc, char *argv[]){
+ if(argc != 3){
+ return RESULT_SHOWUSAGE;
+ }
+ rtcpstats = 1;
+ ast_cli(fd, "RTCP Stats Enabled\n");
+ return RESULT_SUCCESS;
+}
+
static int rtp_no_debug(int fd, int argc, char *argv[])
{
if(argc !=3)
@@ -1867,6 +2470,25 @@
return RESULT_SUCCESS;
}
+static int rtcp_no_debug(int fd, int argc, char *argv[])
+{
+ if(argc !=4)
+ return RESULT_SHOWUSAGE;
+ rtcpdebug = 0;
+ ast_cli(fd,"RTCP Debugging Disabled\n");
+ return RESULT_SUCCESS;
+}
+
+static int rtcp_no_stats(int fd, int argc, char *argv[])
+{
+ if(argc !=4)
+ return RESULT_SHOWUSAGE;
+ rtcpstats = 0;
+ ast_cli(fd,"RTCP Stats Disabled\n");
+ return RESULT_SUCCESS;
+}
+
+
static char debug_usage[] =
"Usage: rtp debug [ip host[:port]]\n"
" Enable dumping of all RTP packets to and from host.\n";
@@ -1883,6 +2505,37 @@
static struct ast_cli_entry cli_no_debug =
{{ "rtp", "no", "debug", NULL } , rtp_no_debug, "Disable RTP debugging", no_debug_usage };
+
+static char rtcp_debug_usage[] =
+ "Usage: rtp rtcp debug [ip host[:port]]\n"
+ " Enable dumping of all RTCP packets to and from host.\n";
+
+static char rtcp_no_debug_usage[] =
+ "Usage: rtp rtcp no debug\n"
+ " Disable all RTCP debugging\n";
+
+static char rtcp_stats_usage[] =
+ "Usage: rtp rtcp stats\n"
+ " Enable dumping of RTCP stats.\n";
+
+static char rtcp_no_stats_usage[] =
+ "Usage: rtp rtcp no stats\n"
+ " Disable all RTCP stats\n";
+
+static struct ast_cli_entry cli_debug_ip_rtcp =
+{{ "rtp", "rtcp", "debug", "ip", NULL } , rtcp_do_debug, "Enable RTCP debugging on IP", rtcp_debug_usage };
+
+static struct ast_cli_entry cli_debug_rtcp =
+{{ "rtp", "rtcp", "debug", NULL } , rtcp_do_debug, "Enable RTCP debugging", rtcp_debug_usage };
+
+static struct ast_cli_entry cli_no_debug_rtcp =
+{{ "rtp", "rtcp", "no", "debug", NULL } , rtcp_no_debug, "Disable RTCP debugging", rtcp_no_debug_usage };
+
+static struct ast_cli_entry cli_stats_rtcp =
+{{ "rtp", "rtcp", "stats", NULL } , rtcp_do_stats, "Enable RTCP stats", rtcp_stats_usage };
+
+static struct ast_cli_entry cli_no_stats_rtcp =
+{{ "rtp", "rtcp", "no", "stats", NULL } , rtcp_no_stats, "Disable RTCP stats", rtcp_no_stats_usage };
int ast_rtp_reload(void)
{
@@ -1908,6 +2561,15 @@
if (rtpend > 65535)
rtpend = 65535;
}
+ if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) {
+ rtcpinterval = atoi(s);
+ if (rtcpinterval == 0)
+ rtcpinterval = 0; /* Just so we're clear... it's zero */
+ if (rtcpinterval < RTCP_MIN_INTERVALMS)
+ rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */
+ if (rtcpinterval > RTCP_MAX_INTERVALMS)
+ rtcpinterval = RTCP_MAX_INTERVALMS;
+ }
if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) {
#ifdef SO_NO_CHECK
if (ast_false(s))
@@ -1945,5 +2607,14 @@
ast_cli_register(&cli_debug);
ast_cli_register(&cli_debug_ip);
ast_cli_register(&cli_no_debug);
+
+ ast_cli_register(&cli_debug_rtcp);
+ ast_cli_register(&cli_debug_ip_rtcp);
+ ast_cli_register(&cli_no_debug_rtcp);
+
+ ast_cli_register(&cli_stats_rtcp);
+ ast_cli_register(&cli_no_stats_rtcp);
+
ast_rtp_reload();
}
+
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