Exploring the Mediastreamer2 VoIP engine. Part 6

The material of the article is taken from my zen channel.

Audio signal transmission via RTP stream

In the past article we have assembled a remote control circuit from a generator and a tone detector that work within the same program. In this article, we will learn how to use the RTP protocol (RFC 3550 - RTP: A Transport Protocol for Real-Time Applications) for receiving/transmitting an audio signal via an Ethernet network.

RTP protocol (Real Time Protocol) in translation means a real-time protocol, it is used to transmit sound, video, data, everything that requires real-time transmission. Let's take an audio signal as an example. The flexibility of the protocol is such that it allows you to transmit an audio signal with a predetermined quality.

The transmission is done using UDP packets, which means that packet loss is quite acceptable during transmission. Each packet is embedded with a special RTP header and a data block of the transmitted signal. The header contains a randomly selected signal source identifier, information about the type of transmitted signal, a unique packet sequence number so that the packets can be arranged in the correct order during decoding, regardless of the order in which they were delivered by the network. The header may also contain additional information, the so-called extension, which allows the header to be adapted to the application in a particular application.

The data block contains the payload of the packet. The internal organization of the content depends on the type of load, it can be samples of a monophonic signal, a stereo signal, a video image line, etc.

The load type is indicated by a seven-bit number. RFC3551 (RTP Profile for Audio and Video Conferenceswith Minimal Control) sets several types of load in the corresponding table, a description of the types of load and the meaning of the codes by which they are indicated are given. Some of the codes are not strictly tied to any type of load - they can be used to indicate an arbitrary load.

The size of a data block is limited from above by the maximum packet size that can be transmitted on a given network without segmentation (the MTU parameter). In general, this is no more than 1500 bytes. Thus, in order to increase the amount of data transmitted per second, you can increase the packet size up to a certain point, and then you will need to increase the frequency of sending packets. In the media streamer, this is a configurable setting. By default, it is 50 Hz, i.e. 50 packets per second. The sequence of transmitted RTP packets will be called an RTP stream.

To start transferring data between the source and the receiver, it is enough that the transmitter knows the IP address of the receiver and the port number that it uses to receive. Those. without any preliminary procedures, the source begins to transmit data, and the receiver, in turn, is ready to immediately receive and process them. According to the standard, the port number used to transmit or receive an RTP stream must be even.

In situations where it is impossible to know the address of the receiver in advance, servers are used on which the receivers leave their address, and the transmitter can request it, referring to some unique receiver name.

In cases where the quality of the communication channel or the capabilities of the receiver are unknown, a feedback channel is organized through which the receiver can inform the transmitter about its capabilities, the number of packets that it missed, etc. This channel uses the RTCP protocol. The format of packets transmitted in this channel is defined in RFC 3605. Relatively little data is transmitted over this channel 200..300 bytes per second, so in general, its presence is not burdensome. The port number to which RTCP packets are sent must be odd and one greater than the port number from which the RTP stream comes. In our example, we will not use this channel, since the capabilities of the receiver and channel obviously exceed our modest needs.

In our program, the data transmission scheme, unlike the scheme of the previous example, will be divided into two parts: a transmit path and a receive path. For each part, we will make our own clock source, as shown in the header image.

One-way communication between them will be carried out using the RTP protocol. In this example, we do not need an external network, since both the transmitter and receiver will be located on the same computer - the packets will go inside it.

The media streamer uses two filters to establish an RTP stream: MS_RTP_SEND and MS_RTP_RECV. The first performs the transmission of the second reception of the RTP stream. For these filters to work, they need to pass a pointer to an RTP session object, which can either convert a stream of data blocks into a stream of RTP packets or perform the opposite action. Since the internal data format of the media streamer does not match the data format of the RTP packet, before sending data to MS_RTP_SEND, you need to use a converter filter (encoder), which converts 16-bit samples of the audio signal into eight-bit samples encoded according to the u-law (mu-law). On the receiving side, the reverse function is performed by the decoder filter.

Below is the text of the program that implements the scheme shown in the figure (the # symbols before the include directives are removed, do not forget to put them):

/* Файл mstest6.c Имитатор пульта управления и приемника. */
#include <mediastreamer2/msfilter.h>
#include <mediastreamer2/msticker.h>
#include <mediastreamer2/dtmfgen.h>
#include <mediastreamer2/mssndcard.h>
#include <mediastreamer2/msvolume.h>
#include <mediastreamer2/mstonedetector.h>
#include <mediastreamer2/msrtp.h>
#include <ortp/rtpsession.h>
#include <ortp/payloadtype.h>
/* Подключаем заголовочный файл с функциями управления событиями
* медиастримера.*/
include <mediastreamer2/mseventqueue.h>
#define PCMU 0
/* Функция обратного вызова, она будет вызвана фильтром, как только он
обнаружит совпадение характеристик входного сигнала с заданными. */
static void tone_detected_cb(void *data, MSFilter *f, unsigned int event_id,
MSToneDetectorEvent *ev)
{
printf("Принята команда: %sn", ev->tone_name);
}
/*----------------------------------------------------------------------------*/
/* Функция регистрации типов полезных нагрузок. */
void register_payloads(void)
{
/*Регистрируем типы нагрузок в таблице профилей. Позднее, по индексу
взятому из заголовка RTP-пакета из этой таблицы будут извлекаться
параметры нагрузки, необходимые для декодирования данных пакета. */
rtp_profile_set_payload (&av_profile, PCMU, &payload_type_pcm8000);
}
/*----------------------------------------------------------------------------*/
/* Эта функция создана из функции create_duplex_rtpsession() в audiostream.c
медиастримера2. */
static RtpSession *
create_rtpsession (int loc_rtp_port, int loc_rtcp_port,
bool_t ipv6, RtpSessionMode mode)
{
RtpSession *rtpr;
rtpr = rtp_session_new ((int) mode);
rtp_session_set_scheduling_mode (rtpr, 0);
rtp_session_set_blocking_mode (rtpr, 0);
rtp_session_enable_adaptive_jitter_compensation (rtpr, TRUE);
rtp_session_set_symmetric_rtp (rtpr, TRUE);
rtp_session_set_local_addr (rtpr, ipv6 ? "::" : "0.0.0.0", loc_rtp_port,
loc_rtcp_port);
rtp_session_signal_connect (rtpr, "timestamp_jump",
(RtpCallback) rtp_session_resync, 0);
rtp_session_signal_connect (rtpr, "ssrc_changed",
(RtpCallback) rtp_session_resync, 0);
rtp_session_set_ssrc_changed_threshold (rtpr, 0);
rtp_session_set_send_payload_type(rtpr, PCMU);
/* По умолчанию выключаем RTCP-сессию, так как наш пульт не будет использовать её. */
rtp_session_enable_rtcp (rtpr, FALSE);
return rtpr;
}
/*----------------------------------------------------------------------------*/
int main()
{
ms_init();
/* Создаем экземпляры фильтров. */
MSFilter *voidsource = ms_filter_new(MS_VOID_SOURCE_ID);
MSFilter *dtmfgen = ms_filter_new(MS_DTMF_GEN_ID);
MSFilter *volume = ms_filter_new(MS_VOLUME_ID);
MSSndCard *card_playback =
ms_snd_card_manager_get_default_card(ms_snd_card_manager_get());
MSFilter *snd_card_write = ms_snd_card_create_writer(card_playback);
MSFilter *detector = ms_filter_new(MS_TONE_DETECTOR_ID);
/* Очищаем массив находящийся внутри детектора тонов, он описывает
* особые приметы разыскиваемых сигналов.*/
ms_filter_call_method(detector, MS_TONE_DETECTOR_CLEAR_SCANS, 0);
/* Подключаем к фильтру функцию обратного вызова. */
ms_filter_set_notify_callback(detector,
(MSFilterNotifyFunc)tone_detected_cb, NULL);
/* Создаем массив, каждый элемент которого описывает характеристику
* одного из тонов, который требуется обнаруживать: Текстовое имя
* данного элемента, частота в герцах, длительность в миллисекундах,
* минимальный уровень относительно 0,775В. */
MSToneDetectorDef scan[6]=
{
{"V+",440, 100, 0.1}, /* Команда "Увеличить громкость". */
{"V-",540, 100, 0.1}, /* Команда "Уменьшить громкость". */
{"C+",640, 100, 0.1}, /* Команда "Увеличить номер канала". */
{"C-",740, 100, 0.1}, /* Команда "Уменьшить номер канала". */
{"ON",840, 100, 0.1}, /* Команда "Включить телевизор". */
{"OFF", 940, 100, 0.1}/* Команда "Выключить телевизор". */
};
/* Передаем "приметы" сигналов детектор тонов. */
int i;
for (i = 0; i < 6; i++)
{
ms_filter_call_method(detector, MS_TONE_DETECTOR_ADD_SCAN,
&scan[i]);
}
/* Создаем фильтры кодера и декодера */
MSFilter *encoder = ms_filter_create_encoder("PCMU");
MSFilter *decoder=ms_filter_create_decoder("PCMU");
/* Регистрируем типы нагрузки. */
register_payloads();
/* Создаем RTP-сессию передатчика. */
RtpSession *tx_rtp_session = create_rtpsession (8010, 8011, FALSE, RTP_SESSION_SENDONLY);
rtp_session_set_remote_addr_and_port(tx_rtp_session,"127.0.0.1", 7010, 7011);
rtp_session_set_send_payload_type(tx_rtp_session, PCMU);
MSFilter *rtpsend = ms_filter_new(MS_RTP_SEND_ID);
ms_filter_call_method(rtpsend, MS_RTP_SEND_SET_SESSION, tx_rtp_session);
/* Создаем RTP-сессию приемника. */
MSFilter *rtprecv = ms_filter_new(MS_RTP_RECV_ID);
RtpSession *rx_rtp_session = create_rtpsession (7010, 7011, FALSE, RTP_SESSION_RECVONLY);
ms_filter_call_method(rtprecv, MS_RTP_RECV_SET_SESSION, rx_rtp_session);
/* Создаем источники тактов - тикеры. */
MSTicker *ticker_tx = ms_ticker_new();
MSTicker *ticker_rx = ms_ticker_new();
/* Соединяем фильтры передатчика. */
ms_filter_link(voidsource, 0, dtmfgen, 0);
ms_filter_link(dtmfgen, 0, volume, 0);
ms_filter_link(volume, 0, encoder, 0);
ms_filter_link(encoder, 0, rtpsend, 0);
/* Соединяем фильтры приёмника. */
ms_filter_link(rtprecv, 0, decoder, 0);
ms_filter_link(decoder, 0, detector, 0);
ms_filter_link(detector, 0, snd_card_write, 0);
/* Подключаем источник тактов. */
ms_ticker_attach(ticker_tx, voidsource);
ms_ticker_attach(ticker_rx, rtprecv);
/* Настраиваем структуру, управляющую выходным сигналом генератора. */
MSDtmfGenCustomTone dtmf_cfg;
dtmf_cfg.tone_name[0] = 0;
dtmf_cfg.duration = 1000;
dtmf_cfg.frequencies[0] = 440;
/* Будем генерировать один тон, частоту второго тона установим в 0. */
dtmf_cfg.frequencies[1] = 0;
dtmf_cfg.amplitude = 1.0;
dtmf_cfg.interval = 0.;
dtmf_cfg.repeat_count = 0.;
/* Организуем цикл сканирования нажатых клавиш. Ввод нуля завершает
* цикл и работу программы. */
char key='9';
printf("Нажмите клавишу команды, затем ввод.n"
"Для завершения программы введите 0.n");
while(key != '0')
{
key = getchar();
if ((key >= 49) && (key <= 54))
{
printf("Отправлена команда: %cn", key);
/* Устанавливаем частоту генератора в соответствии с
* кодом нажатой клавиши. */
dtmf_cfg.frequencies[0] = 440 + 100*(key-49);
/* Включаем звуковой генератор c обновленной частотой. */
ms_filter_call_method(dtmfgen, MS_DTMF_GEN_PLAY_CUSTOM,
(void*)&dtmf_cfg);
}
/* Укладываем тред в спячку на 20мс, чтобы другие треды
* приложения получили время на работу. */
ms_usleep(20000);
}
}

Compile, run. The program will work as in the previous example, but the data will be transmitted via the RTP stream.

In the next article, we will split this program into two independent applications - a receiver and a transmitter, and run them in different terminals. In parallel, we will learn how to analyze RTP packets using the TShark program.

Source: habr.com