Files
UltraGrid/src/rtsp/BasicRTSPOnlySubsession.cpp
Martin Pulec 1ad7722a67 RTSP: support stream redirect
When the client doesn't call TEARDOWN (like ffplay doesn't),
the stream could not have been played until the timeout (given by
`reclamationTestSeconds`). After that (or when TEARDOWN was called),
`BasicRTSPOnlySubsession::deleteStream()` is called allowing the new
stream.

After this change, the stream can be redirected withot explicit TEARDOWN
or timeout.
2024-01-09 16:20:18 +01:00

358 lines
12 KiB
C++

/*
* FILE: rtsp/BasicRTSPOnlySubsession.cpp
* AUTHORS: David Cassany <david.cassany@i2cat.net>
* Gerard Castillo <gerard.castillo@i2cat.net>
*
* Copyright (c) 2005-2010 Fundació i2CAT, Internet I Innovació Digital a Catalunya
* Copyright (c) 2014-2023 CESNET, z. s. p. o.
*
* Redistribution and use in source and binary forms, with or without
* modification, is permitted provided that the following conditions
* are met:
*
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* 3. All advertising materials mentioning features or use of this software
* must display the following acknowledgement:
*
* This product includes software developed by the Fundació i2CAT,
* Internet I Innovació Digital a Catalunya. This product also includes
* software developed by CESNET z.s.p.o.
*
* 4. Neither the name of the University nor of the Institute may be used
* to endorse or promote products derived from this software without
* specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHORS AND CONTRIBUTORS
* "AS IS" AND ANY EXPRESSED OR IMPLIED WARRANTIES, INCLUDING,
* BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY
* AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHORS OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT,
* INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
* SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE,
* EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*
*/
#include "rtsp/BasicRTSPOnlySubsession.hh"
#include <BasicUsageEnvironment.hh>
#include <RTSPServer.hh>
#include <GroupsockHelper.hh>
#include "messaging.h"
BasicRTSPOnlySubsession*
BasicRTSPOnlySubsession::createNew(UsageEnvironment& env,
Boolean reuseFirstSource, struct module *mod, rtps_types_t avType,
audio_codec_t audio_codec, int audio_sample_rate, int audio_channels,
int audio_bps, int rtp_port, int rtp_port_audio) {
return new BasicRTSPOnlySubsession(env, reuseFirstSource, mod, avType,
audio_codec, audio_sample_rate, audio_channels, audio_bps, rtp_port, rtp_port_audio);
}
BasicRTSPOnlySubsession::BasicRTSPOnlySubsession(UsageEnvironment& env,
Boolean reuseFirstSource, struct module *mod, rtps_types_t avType,
audio_codec_t audio_codec, int audio_sample_rate, int audio_channels,
int audio_bps, int rtp_port, int rtp_port_audio) :
ServerMediaSubsession(env), fSDPLines(NULL), fReuseFirstSource(
reuseFirstSource), fLastStreamToken(NULL) {
Vdestination = NULL;
Adestination = NULL;
gethostname(fCNAME, sizeof fCNAME);
this->fmod = mod;
this->avType = avType;
this->audio_codec = audio_codec;
this->audio_sample_rate = audio_sample_rate;
this->audio_channels = audio_channels;
this->audio_bps = audio_bps;
this->rtp_port = rtp_port;
this->rtp_port_audio = rtp_port_audio;
fCNAME[sizeof fCNAME - 1] = '\0';
}
BasicRTSPOnlySubsession::~BasicRTSPOnlySubsession() {
delete[] fSDPLines;
delete Adestination;
delete Vdestination;
}
char const* BasicRTSPOnlySubsession::sdpLines() {
if (fSDPLines == NULL) {
setSDPLines();
}
return fSDPLines;
}
void BasicRTSPOnlySubsession::setSDPLines() {
//TODO: should be more dynamic
//VStream
if (avType == video || avType == av) {
unsigned estBitrate = 5000;
char const* mediaType = "video";
uint8_t rtpPayloadType = 96;
AddressString ipAddressStr(fServerAddressForSDP);
char* rtpmapLine = strdup("a=rtpmap:96 H264/90000\n");
//char const* auxSDPLine = "";
char const* const sdpFmt = "m=%s %u RTP/AVP %u\r\n"
"c=IN IP4 %s\r\n"
"b=AS:%u\r\n"
"a=rtcp:%d\r\n"
"%s"
"a=control:%s\r\n";
unsigned sdpFmtSize = strlen(sdpFmt) + strlen(mediaType) + 5 /* max short len */
+ 3 /* max char len */
+ strlen(ipAddressStr.val()) + 20 /* max int len */
+ strlen(rtpmapLine) + strlen(trackId());
char* sdpLines = new char[sdpFmtSize];
snprintf(sdpLines, sdpFmtSize, sdpFmt, mediaType, // m= <media>
rtp_port,//fPortNumForSDP, // m= <port>
rtpPayloadType, // m= <fmt list>
ipAddressStr.val(), // c= address
estBitrate, // b=AS:<bandwidth>
rtp_port + 1,
rtpmapLine, // a=rtpmap:... (if present)
trackId()); // a=control:<track-id>
fSDPLines = sdpLines;
free(rtpmapLine);
}
//AStream
if (avType == audio || avType == av) {
unsigned estBitrate = 384;
char const* mediaType = "audio";
AddressString ipAddressStr(fServerAddressForSDP);
uint8_t rtpPayloadType;
if (audio_sample_rate == 8000 && audio_channels == 1) { //NOW NOT COMPUTING 1 BPS BECAUSE RESAMPLER FORCES TO 2 BPS...
if (audio_codec == AC_MULAW)
rtpPayloadType = 0;
else if (audio_codec == AC_ALAW)
rtpPayloadType = 8;
else rtpPayloadType = 97;
} else {
rtpPayloadType = 97;
}
char* rtpmapLine = strdup("a=rtpmap:97 PCMU/48000/2\n"); //only to alloc max possible size
//char const* auxSDPLine = "";
const int sdp_ch_count = audio_codec == AC_OPUS ? 2 :
audio_channels; // RFC 7587 enforces 2 for Opus
char const* const sdpFmt = "m=%s %u RTP/AVP %u\r\n"
"c=IN IP4 %s\r\n"
"b=AS:%u\r\n"
"a=rtcp:%d\r\n"
"a=rtpmap:%u %s/%d/%d\r\n"
"a=control:%s\r\n";
unsigned sdpFmtSize = strlen(sdpFmt) + strlen(mediaType) + 5 /* max short len */
+ 3 /* max char len */
+ strlen(ipAddressStr.val()) + 20 /* max int len */
+ strlen(rtpmapLine) + strlen(trackId());
char* sdpLines = new char[sdpFmtSize];
snprintf(sdpLines, sdpFmtSize, sdpFmt,
mediaType, // m= <media>
rtp_port_audio,//fPortNumForSDP, // m= <port>
rtpPayloadType, // m= <fmt list>
ipAddressStr.val(), // c= address
estBitrate, // b=AS:<bandwidth>
//rtpmapLine, // a=rtpmap:... (if present)
rtp_port_audio + 1,
rtpPayloadType,
audio_codec == AC_MULAW ? "PCMU" : audio_codec == AC_ALAW ? "PCMA" : "opus",
audio_sample_rate,
sdp_ch_count,
trackId()); // a=control:<track-id>
fSDPLines = sdpLines;
free(rtpmapLine);
}
}
void BasicRTSPOnlySubsession::getStreamParameters(unsigned /* clientSessionId */,
netAddressBits clientAddress, Port const& clientRTPPort,
Port const& clientRTCPPort, int /* tcpSocketNum */,
unsigned char /* rtpChannelId */, unsigned char /* rtcpChannelId */,
netAddressBits& destinationAddress, uint8_t& /*destinationTTL*/,
Boolean& /* isMulticast */, Port& serverRTPPort, Port& serverRTCPPort,
void*& /* streamToken */) {
if (avType == video || avType == av) {
Port rtp(rtp_port);
serverRTPPort = rtp;
Port rtcp(rtp_port + 1);
serverRTCPPort = rtcp;
if (fSDPLines == NULL) {
setSDPLines();
}
if (destinationAddress == 0) {
destinationAddress = clientAddress;
}
struct in_addr destinationAddr;
destinationAddr.s_addr = destinationAddress;
delete Vdestination;
Vdestination = new Destinations(destinationAddr, clientRTPPort,
clientRTCPPort);
}
if (avType == audio || avType == av) {
Port rtp(rtp_port_audio);
serverRTPPort = rtp;
Port rtcp(rtp_port_audio + 1);
serverRTCPPort = rtcp;
if (fSDPLines == NULL) {
setSDPLines();
}
if (destinationAddress == 0) {
destinationAddress = clientAddress;
}
struct in_addr destinationAddr;
destinationAddr.s_addr = destinationAddress;
delete Adestination;
Adestination = new Destinations(destinationAddr, clientRTPPort,
clientRTCPPort);
}
}
void BasicRTSPOnlySubsession::startStream(unsigned /* clientSessionId */,
void* /* streamToken */, TaskFunc* /* rtcpRRHandler */,
void* /* rtcpRRHandlerClientData */, unsigned short& /* rtpSeqNum */,
unsigned& /* rtpTimestamp */,
ServerRequestAlternativeByteHandler* /* serverRequestAlternativeByteHandler */,
void* /* serverRequestAlternativeByteHandlerClientData */) {
struct response *resp = NULL;
if (Vdestination != NULL) {
if (avType == video || avType == av) {
char pathV[1024];
memset(pathV, 0, sizeof(pathV));
enum module_class path_sender[] = { MODULE_CLASS_SENDER,
MODULE_CLASS_NONE };
append_message_path(pathV, sizeof(pathV), path_sender);
//CHANGE DST PORT
struct msg_sender *msgV1 = (struct msg_sender *) new_message(
sizeof(struct msg_sender));
msgV1->tx_port = ntohs(Vdestination->rtpPort.num());
msgV1->type = SENDER_MSG_CHANGE_PORT;
resp = send_message(fmod, pathV, (struct message *) msgV1);
free_response(resp);
//CHANGE DST ADDRESS
struct msg_sender *msgV2 = (struct msg_sender *) new_message(
sizeof(struct msg_sender));
strncpy(msgV2->receiver, inet_ntoa(Vdestination->addr),
sizeof(msgV2->receiver) - 1);
msgV2->type = SENDER_MSG_CHANGE_RECEIVER;
resp = send_message(fmod, pathV, (struct message *) msgV2);
free_response(resp);
}
}
if (Adestination != NULL) {
if (avType == audio || avType == av) {
char pathA[1024];
memset(pathA, 0, sizeof(pathA));
enum module_class path_sender[] = { MODULE_CLASS_AUDIO,
MODULE_CLASS_SENDER, MODULE_CLASS_NONE };
append_message_path(pathA, sizeof(pathA), path_sender);
//CHANGE DST PORT
struct msg_sender *msgA1 = (struct msg_sender *) new_message(
sizeof(struct msg_sender));
msgA1->tx_port = ntohs(Adestination->rtpPort.num());
msgA1->type = SENDER_MSG_CHANGE_PORT;
resp = send_message(fmod, pathA, (struct message *) msgA1);
free_response(resp);
resp = NULL;
//CHANGE DST ADDRESS
struct msg_sender *msgA2 = (struct msg_sender *) new_message(
sizeof(struct msg_sender));
strncpy(msgA2->receiver, inet_ntoa(Adestination->addr),
sizeof(msgA2->receiver) - 1);
msgA2->type = SENDER_MSG_CHANGE_RECEIVER;
resp = send_message(fmod, pathA, (struct message *) msgA2);
free_response(resp);
resp = NULL;
}
}
}
void BasicRTSPOnlySubsession::deleteStream(unsigned /* clientSessionId */,
void*& /* streamToken */) {
if (Vdestination != NULL) {
if (avType == video || avType == av) {
char pathV[1024];
Vdestination = NULL;
memset(pathV, 0, sizeof(pathV));
enum module_class path_sender[] = { MODULE_CLASS_SENDER,
MODULE_CLASS_NONE };
append_message_path(pathV, sizeof(pathV), path_sender);
//CHANGE DST PORT
struct msg_sender *msgV1 = (struct msg_sender *) new_message(
sizeof(struct msg_sender));
msgV1->tx_port = rtp_port;
msgV1->type = SENDER_MSG_CHANGE_PORT;
struct response *resp;
resp = send_message(fmod, pathV, (struct message *) msgV1);
free_response(resp);
//CHANGE DST ADDRESS
struct msg_sender *msgV2 = (struct msg_sender *) new_message(
sizeof(struct msg_sender));
strncpy(msgV2->receiver, "127.0.0.1", sizeof(msgV2->receiver) - 1);
msgV2->type = SENDER_MSG_CHANGE_RECEIVER;
resp = send_message(fmod, pathV, (struct message *) msgV2);
free_response(resp);
}
}
if (Adestination != NULL) {
if (avType == audio || avType == av) {
char pathA[1024];
Adestination = NULL;
memset(pathA, 0, sizeof(pathA));
enum module_class path_sender[] = { MODULE_CLASS_AUDIO,
MODULE_CLASS_SENDER, MODULE_CLASS_NONE };
append_message_path(pathA, sizeof(pathA), path_sender);
//CHANGE DST PORT
struct msg_sender *msgA1 = (struct msg_sender *) new_message(
sizeof(struct msg_sender));
//TODO: GET AUDIO PORT SET (NOT A COMMON CASE WHEN RTSP IS ENABLED: DEFAULT -> vport + 2)
msgA1->tx_port = rtp_port_audio;
msgA1->type = SENDER_MSG_CHANGE_PORT;
struct response *resp;
resp = send_message(fmod, pathA, (struct message *) msgA1);
free_response(resp);
//CHANGE DST ADDRESS
struct msg_sender *msgA2 = (struct msg_sender *) new_message(
sizeof(struct msg_sender));
strncpy(msgA2->receiver, "127.0.0.1", sizeof(msgA2->receiver) - 1);
msgA2->type = SENDER_MSG_CHANGE_RECEIVER;
resp = send_message(fmod, pathA, (struct message *) msgA2);
free_response(resp);
}
}
}
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