From 9befc6968493c84e90a25d4b02889c8ca1853cc8 Mon Sep 17 00:00:00 2001
From: Mark Spencer <markster@digium.com>
Date: Sat, 28 Jun 2003 18:35:58 +0000
Subject: [PATCH] Properly handle broken up packets
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1129 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_sip.c | 7 ++--
file.c | 2 +-
rtp.c | 90 ++++++++++++++-------------------------------
3 files changed, 32 insertions(+), 67 deletions(-)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 35a506aeb3..e911962c38 100755
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1604,7 +1604,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
int portno=0;
int vportno=0;
int peercapability, peernoncodeccapability;
- int vpeercapability, vpeernoncodeccapability;
+ int vpeercapability=0, vpeernoncodeccapability=0;
struct sockaddr_in sin;
char *codecs;
struct hostent *hp;
@@ -1690,8 +1690,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
sdpLineNum_iterator_init(&iterator);
while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
char* mimeSubtype = ast_strdupa(a); // ensures we have enough space
- if (sipdebug)
- ast_verbose("Pre-Found description format %s\n", mimeSubtype);
if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2) continue;
if (sipdebug)
ast_verbose("Found description format %s\n", mimeSubtype);
@@ -1704,7 +1702,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
// Now gather all of the codecs that were asked for:
ast_rtp_get_current_formats(p->rtp,
&peercapability, &peernoncodeccapability);
- ast_rtp_get_current_formats(p->vrtp,
+ if (p->vrtp)
+ ast_rtp_get_current_formats(p->vrtp,
&vpeercapability, &vpeernoncodeccapability);
p->capability = capability & (peercapability | vpeercapability);
p->noncodeccapability = noncodeccapability & (peernoncodeccapability | vpeernoncodeccapability);
diff --git a/file.c b/file.c
index f7a6049645..a38633d8aa 100755
--- a/file.c
+++ b/file.c
@@ -1,4 +1,4 @@
-/*m
+/*
* Asterisk -- A telephony toolkit for Linux.
*
* Generic File Format Support.
diff --git a/rtp.c b/rtp.c
index 6fd6f71903..a60e9d5388 100755
--- a/rtp.c
+++ b/rtp.c
@@ -60,7 +60,6 @@ struct ast_rtp {
char resp;
struct ast_frame f;
unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET];
- int readsofar;
unsigned int ssrc;
unsigned int lastts;
unsigned int lastrxts;
@@ -335,19 +334,17 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
int mark;
unsigned int timestamp;
unsigned int *rtpheader;
- unsigned char cache[12];
static struct ast_frame *f, null_frame = { AST_FRAME_NULL, };
struct rtpPayloadType rtpPT;
len = sizeof(sin);
/* Cache where the header will go */
- memcpy(cache, rtp->rawdata + AST_FRIENDLY_OFFSET + rtp->readsofar, 12);
- res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET + rtp->readsofar, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET - rtp->readsofar,
+ res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET,
0, (struct sockaddr *)&sin, &len);
- rtpheader = (unsigned int *)(rtp->rawdata + rtp->readsofar + AST_FRIENDLY_OFFSET);
+ rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
if (res < 0) {
ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno));
if (errno == EBADF)
@@ -373,18 +370,11 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
seqno &= 0xffff;
timestamp = ntohl(rtpheader[1]);
- /* Restore original data if important */
- if (rtp->readsofar)
- memcpy(rtp->rawdata + AST_FRIENDLY_OFFSET + rtp->readsofar, cache, 12);
- rtpheader = NULL;
-
- rtp->readsofar += (res - hdrlen);
#if 0
printf("Got RTP packet from %s:%d (type %d, seq %d, ts %d, len = %d)\n", inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
#endif
rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
if (!rtpPT.isAstFormat) {
- rtp->readsofar = 0;
// This is special in-band data that's not one of our codecs
if (rtpPT.code == AST_RTP_DTMF) {
/* It's special -- rfc2833 process it */
@@ -429,16 +419,14 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
/* Send any pending DTMF */
if (rtp->resp && !rtp->dtmfcount) {
- rtp->readsofar = 0;
ast_log(LOG_DEBUG, "Sending pending DTMF\n");
return send_dtmf(rtp);
}
rtp->f.mallocd = 0;
- rtp->f.datalen = rtp->readsofar;
+ rtp->f.datalen = res - hdrlen;
rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) {
- rtp->readsofar = 0;
switch(rtp->f.subclass) {
case AST_FORMAT_ULAW:
case AST_FORMAT_ALAW:
@@ -474,11 +462,9 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
/* Video -- samples is # of samples vs. 90000 */
rtp->f.samples = timestamp - rtp->lastividtimestamp;
rtp->lastividtimestamp = timestamp;
- /* Return now if it's not the whole frame */
- if (!mark) {
- return &null_frame;
- }
- rtp->readsofar = 0;
+ if (mark)
+ rtp->f.subclass |= 0x1;
+
}
rtp->f.src = "RTP";
return &rtp->f;
@@ -887,7 +873,6 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
int ms;
int pred;
int mark = 0;
- int pos, len;
ms = calc_txstamp(rtp);
/* Default prediction */
@@ -928,20 +913,8 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
rtp->lastts = pred;
else
ast_log(LOG_DEBUG, "Difference is %d, ms is %d\n", abs(rtp->lastts - pred), ms);
- /* Get a pointer to the header */
- rtpheader = (unsigned int *)(f->data - hdrlen);
- rtpheader[0] = htonl((2 << 30) | (codec << 16) | (rtp->seqno++) | (mark << 23));
- rtpheader[1] = htonl(rtp->lastts);
- rtpheader[2] = htonl(rtp->ssrc);
- if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
- res = sendto(rtp->s, (void *)rtpheader, f->datalen + hdrlen, 0, &rtp->them, sizeof(rtp->them));
- if (res <0)
- ast_log(LOG_NOTICE, "RTP Transmission error to %s:%d: %s\n", inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
-#if 0
- printf("Sent %d bytes of RTP data to %s:%d\n", res, inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
-#endif
- }
} else {
+ mark = f->subclass & 0x1;
pred = rtp->lastovidtimestamp + f->samples;
/* Re-calculate last TS */
rtp->lastts = rtp->lastts + ms * 90;
@@ -953,30 +926,19 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
ast_log(LOG_DEBUG, "Difference is %d, ms is %d\n", abs(rtp->lastts - pred), ms);
rtp->lastovidtimestamp = rtp->lastts;
}
- pos = 0;
- while(pos < f->datalen) {
- /* Split packets up if necessary */
- mark = 0;
- len = f->datalen;
- if (len > RTP_MTU)
- len = RTP_MTU;
- if ((pos + len) >= f->datalen)
- mark = 1;
- /* Get a pointer to the header */
- rtpheader = (unsigned int *)(f->data + pos - hdrlen);
- rtpheader[0] = htonl((2 << 30) | (codec << 16) | (rtp->seqno++) | (mark << 23));
- rtpheader[1] = htonl(rtp->lastts);
- rtpheader[2] = htonl(rtp->ssrc);
- if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
- res = sendto(rtp->s, (void *)rtpheader, len + hdrlen, 0, &rtp->them, sizeof(rtp->them));
- if (res <0)
- ast_log(LOG_NOTICE, "RTP Transmission error to %s:%d: %s\n", inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
+ }
+ /* Get a pointer to the header */
+ rtpheader = (unsigned int *)(f->data - hdrlen);
+ rtpheader[0] = htonl((2 << 30) | (codec << 16) | (rtp->seqno++) | (mark << 23));
+ rtpheader[1] = htonl(rtp->lastts);
+ rtpheader[2] = htonl(rtp->ssrc);
+ if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
+ res = sendto(rtp->s, (void *)rtpheader, f->datalen + hdrlen, 0, &rtp->them, sizeof(rtp->them));
+ if (res <0)
+ ast_log(LOG_NOTICE, "RTP Transmission error to %s:%d: %s\n", inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
#if 0
- printf("Sent %d bytes of RTP data to %s:%d\n", res, inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
+ printf("Sent %d bytes of RTP data to %s:%d\n", res, inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
#endif
- }
- pos += len;
- }
}
return 0;
}
@@ -986,6 +948,7 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
struct ast_frame *f;
int codec;
int hdrlen = 12;
+ int subclass;
/* If we have no peer, return immediately */
@@ -1002,24 +965,27 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
return -1;
}
+ subclass = _f->subclass;
+ if (_f->frametype == AST_FRAME_VIDEO)
+ subclass &= ~0x1;
- codec = ast_rtp_lookup_code(rtp, 1, _f->subclass);
+ codec = ast_rtp_lookup_code(rtp, 1, subclass);
if (codec < 0) {
ast_log(LOG_WARNING, "Don't know how to send format %d packets with RTP\n", _f->subclass);
return -1;
}
- if (rtp->lasttxformat != _f->subclass) {
+ if (rtp->lasttxformat != subclass) {
/* New format, reset the smoother */
- ast_log(LOG_DEBUG, "Ooh, format changed from %d to %d\n", rtp->lasttxformat, _f->subclass);
- rtp->lasttxformat = _f->subclass;
+ ast_log(LOG_DEBUG, "Ooh, format changed from %d to %d\n", rtp->lasttxformat, subclass);
+ rtp->lasttxformat = subclass;
if (rtp->smoother)
ast_smoother_free(rtp->smoother);
rtp->smoother = NULL;
}
- switch(_f->subclass) {
+ switch(subclass) {
case AST_FORMAT_ULAW:
case AST_FORMAT_ALAW:
if (!rtp->smoother) {
@@ -1072,7 +1038,7 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
ast_rtp_raw_write(rtp, f, codec);
break;
default:
- ast_log(LOG_WARNING, "Not sure about sending format %d packets\n", _f->subclass);
+ ast_log(LOG_WARNING, "Not sure about sending format %d packets\n", subclass);
// fall through to...
case AST_FORMAT_H261:
case AST_FORMAT_H263:
--
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