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AudioBridge - PlainRTP Participant - Recognize RFC2833 DTMF and publish 'dtmf' event #3538

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AudioBridge - PlainRTP Participant - Recognize RFC2833 DTMF and publish 'dtmf' event #3538

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897099d
Audiobridge, rfc2733 dtmf event is recognized and dtmf event is publi…
keremcadirci Apr 17, 2025
09900a4
Audiobridge, rfc2733 dtmf event updates - #3538
keremcadirci Apr 22, 2025
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104 changes: 102 additions & 2 deletions src/plugins/janus_audiobridge.c
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Original file line number Diff line number Diff line change
Expand Up @@ -946,7 +946,8 @@ room-<unique room ID>: {
"port" : <port you want media to be sent to>,
"payload_type" : <payload type to use for RTP packets (optional; only needed in case Opus is used, automatic for G.711)>,
"audiolevel_ext" : <ID of the audiolevel RTP extension, if used (optional)>,
"fec" : <true|false, whether FEC from Janus to the user should be enabled for the Opus stream (optional; only needed in case Opus is used)>
"fec" : <true|false, whether FEC from Janus to the user should be enabled for the Opus stream (optional; only needed in case Opus is used)>,
"dtmf_pt" : <RTP payload type used for incoming RFC2833 DTMF packets. If set, RTP packets matching dtmf_pt will be emitted as DTMF events (optional)>
}
}
\endverbatim
Expand Down Expand Up @@ -974,6 +975,17 @@ room-<unique room ID>: {
* If you're interested in supporting scenarios where the AudioBridge
* "dials out" (e.g., for outgoing INVITES to SIP endpoints) check the
* \ref aboffer section.
* if a maching rtp with rfc2833 payload type is received the fallowing dtmf event is published.
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@atoppi atoppi May 29, 2025

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typos: start with a capital letter, maching, rtp (packet), fallowing

*
\verbatim
{
"audiobridge" : "event",
"room" : <numeric ID of the room>,
"id" : <unique ID assigned to the participant>,
"dtmf_signal" : "<dtmf signal received, values: 1-9, *, #, A-D>",
"dtmf_duration" : <duration of the dtmf signal>
}
\endverbatim
*
* At this point, whether the participant will be interacting via WebRTC
* or plain RTP, the media-related settings of the participant can be
Expand All @@ -995,6 +1007,14 @@ room-<unique room ID>: {
"record": <true|false, whether to record this user's contribution to a .mjr file (mixer not involved),
"filename": "<basename of the file to record to, -audio.mjr will be added by the plugin; will be relative to mjrs_dir, if configured in the room>",
"group" : "<new group to assign to this participant, if enabled in the room (for forwarding purposes)>"
"rtp" : {
"ip" : "<IP address you want media to be sent to>",
"port" : <port you want media to be sent to>,
"payload_type" : <payload type to use for RTP packets (optional; only needed in case Opus is used, automatic for G.711)>,
"audiolevel_ext" : <ID of the audiolevel RTP extension, if used (optional)>,
"fec" : <true|false, whether FEC should be enabled for the Opus stream (optional; only needed in case Opus is used)>,
"dtmf_pt": <RFC2833 RTP payload type that dtmf signal will be received (optional)>
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@atoppi atoppi May 29, 2025

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please rephrase

}
}
\endverbatim
*
Expand All @@ -1004,7 +1024,9 @@ room-<unique room ID>: {
* to a Janus .mjr file, and \c filename to provide a basename for the path to
* save the file to (notice that this is different from the recording of a whole
* room: this feature only records the packets this user is sending, and is not
* related to the mixer stuff). A successful request will result in a \c ok event:
* related to the mixer stuff). \c rtp is used in case of plain RTP participant
* to set/update remote rtp information while keeping local rtp which is already
* been provided with joined event. A successful request will result in a \c ok event:
*
\verbatim
{
Expand Down Expand Up @@ -1423,6 +1445,7 @@ static struct janus_json_parameter rtp_parameters[] = {
{"payload_type", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
{"audiolevel_ext", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
{"fec", JANUS_JSON_BOOL, 0},
{"dtmf_pt", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
};
static struct janus_json_parameter configure_parameters[] = {
{"muted", JANUS_JSON_BOOL, 0},
Expand Down Expand Up @@ -1735,6 +1758,12 @@ static void janus_audiobridge_file_free(janus_audiobridge_file *ctx) {
}
#endif

/* In case we need to support plain RTP participants, this struct helps with that */
typedef struct janus_audiobridge_plainrtp_dtmf {
uint16_t dtmf_event_id;
uint32_t timestamp;
} janus_audiobridge_plainrtp_dtmf;

/* In case we need to support plain RTP participants, this struct helps with that */
typedef struct janus_audiobridge_plainrtp_media {
char *remote_audio_ip;
Expand All @@ -1748,6 +1777,8 @@ typedef struct janus_audiobridge_plainrtp_media {
int pipefd[2];
GThread *thread;
volatile int initialized;
int dtmf_pt;
janus_audiobridge_plainrtp_dtmf latest_dtmf;
} janus_audiobridge_plainrtp_media;
static void janus_audiobridge_plainrtp_media_cleanup(janus_audiobridge_plainrtp_media *media);
static int janus_audiobridge_plainrtp_allocate_port(janus_audiobridge_plainrtp_media *media);
Expand Down Expand Up @@ -2353,6 +2384,51 @@ static int janus_audiobridge_create_opus_encoder_if_needed(janus_audiobridge_roo
return 0;
}

static uint16_t dtmf_keys[] = {'0', '1', '2', '3', '4', '5', '6', '7', '8', '9', '*', '#', 'A', 'B', 'C', 'D'};
/* Check peer RTP has RFC2833 and push event */
static void janus_audiobridge_send_dtmf_event(janus_audiobridge_participant *participant, char *buffer, int len) {
if(participant->plainrtp_media.dtmf_pt <= 0)
return;
janus_rtp_header *rtp_header = (janus_rtp_header *)buffer;
if(rtp_header->type != participant->plainrtp_media.dtmf_pt)
return;
int plen = 0;
char *payload_buffer = janus_rtp_payload(buffer, len, &plen);
if(plen < 0 || (size_t)plen < sizeof(janus_rtp_rfc2833_payload))
return;
janus_rtp_rfc2833_payload *rfc2833_payload = (janus_rtp_rfc2833_payload *)payload_buffer;
uint16_t duration = ntohs(rfc2833_payload->duration);
if(rfc2833_payload->end == 0)
return;

/* Set up last dtmf to avoid duplication */
if(participant->plainrtp_media.latest_dtmf.dtmf_event_id == rfc2833_payload->event && participant->plainrtp_media.latest_dtmf.timestamp == rtp_header->timestamp)
return;
participant->plainrtp_media.latest_dtmf.dtmf_event_id = rfc2833_payload->event;
participant->plainrtp_media.latest_dtmf.timestamp = rtp_header->timestamp;

/* Parse dtmf key */
uint16_t dtmf_key;
if(rfc2833_payload->event > 15)
return;
dtmf_key = dtmf_keys[rfc2833_payload->event];
char dtmf_key_str[2];
dtmf_key_str[0] = dtmf_key;
dtmf_key_str[1] = '\0';

/* Notify the application */
json_t *info = json_object();
json_object_set_new(info, "audiobridge", json_string("event"));
json_object_set_new(info, "room", string_ids ? json_string(participant->room->room_id_str) : json_integer(participant->room->room_id));
json_object_set_new(info, "id", string_ids ? json_string(participant->user_id_str) : json_integer(participant->user_id));
json_object_set_new(info, "dtmf_signal", json_string(dtmf_key_str));
json_object_set_new(info, "dtmf_duration", json_integer(duration));
int ret = gateway->push_event(participant->session->handle, &janus_audiobridge_plugin, NULL, info, NULL);
JANUS_LOG(LOG_VERB, " >> Pushing event to peer: %d (%s)\n", ret, janus_get_api_error(ret));
json_decref(info);
return;
}

static int janus_audiobridge_create_static_rtp_forwarder(janus_config_category *cat, janus_audiobridge_room *audiobridge) {
guint32 forwarder_id = 0;
janus_config_item *forwarder_id_item = janus_config_get(config, cat, janus_config_type_item, "rtp_forward_id");
Expand Down Expand Up @@ -7036,6 +7112,17 @@ static void *janus_audiobridge_handler(void *data) {
opus_encoder_ctl(participant->encoder, OPUS_SET_INBAND_FEC(participant->fec));
opus_encoder_ctl(participant->encoder, OPUS_SET_PACKET_LOSS_PERC(participant->expected_loss));
}
/* rfc2833 payload type is set */
json_t *dtmf_pt_json = json_object_get(rtp, "dtmf_pt");
if(dtmf_pt_json) {
int dtmf_pt = json_integer_value(dtmf_pt_json);
if(janus_is_dynamic_payload_type(dtmf_pt)) {
participant->plainrtp_media.dtmf_pt = dtmf_pt;
} else {
participant->plainrtp_media.dtmf_pt = -1;
JANUS_LOG(LOG_WARN, "Invalid dtmf_pt %"SCNi32",\n", dtmf_pt);
}
}
/* Create the socket */
janus_mutex_lock(&participant->pmutex);
janus_audiobridge_plainrtp_media_cleanup(&participant->plainrtp_media);
Expand Down Expand Up @@ -7292,6 +7379,17 @@ static void *janus_audiobridge_handler(void *data) {
const char *ip = json_string_value(json_object_get(rtp, "ip"));
uint16_t port = json_integer_value(json_object_get(rtp, "port"));
janus_mutex_lock(&participant->pmutex);
/* rfc2833 payload type is set */
json_t *dtmf_pt_json = json_object_get(rtp, "dtmf_pt");
if(dtmf_pt_json) {
int dtmf_pt = json_integer_value(dtmf_pt_json);
if(janus_is_dynamic_payload_type(dtmf_pt)) {
participant->plainrtp_media.dtmf_pt = dtmf_pt;
} else {
participant->plainrtp_media.dtmf_pt = -1;
JANUS_LOG(LOG_WARN, "Invalid dtmf_pt %"SCNi32",\n", dtmf_pt);
}
}
if(ip == NULL || port == 0) {
JANUS_LOG(LOG_ERR, "[AudioBridge-%p] Can't connect socket, no remote address provided\n", session);
} else {
Expand Down Expand Up @@ -9586,6 +9684,8 @@ static void *janus_audiobridge_plainrtp_relay_thread(void *data) {
/* Handle as a WebRTC RTP packet */
packet.length = bytes;
janus_audiobridge_incoming_rtp(session->handle, &packet);
/* Handle rtp if rfc2833 event*/
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@atoppi atoppi Apr 22, 2025

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I'm not sure that parsing and handling RFC2833 packets in janus_audiobridge_plainrtp_relay_thread is a good idea.
We postpone the inbound packets processing in janus_audiobridge_participant_thread, where packets are being fetched from the jitter buffer.

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@keremcadirci keremcadirci Apr 22, 2025

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Unfortunately, it seems a little tough for me to solve.
RTP packets other than Opus or G.711 are not directed to, and are not received by, the janus_audiobridge_participant_thread.
The janus_audiobridge_participant_thread gets RTP packets from the jitter buffer based on length:
ret = jitter_buffer_get(participant->jitter, &jbp, participant->codec == JANUS_AUDIOCODEC_OPUS ? 960 : 160, NULL);

Any suggestions would be welcome

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@lminiero lminiero Apr 23, 2025

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That's not the length, it's the expected timestamp jump.

janus_audiobridge_send_dtmf_event(participant, buffer, bytes);
continue;
}
}
Expand Down
7 changes: 7 additions & 0 deletions src/rtp.c
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Original file line number Diff line number Diff line change
Expand Up @@ -35,6 +35,13 @@ gboolean janus_is_rtp(char *buf, guint len) {
return ((header->type < 64) || (header->type >= 96));
}

gboolean janus_is_dynamic_payload_type(int payload_type){
if (96 >= payload_type || 127 <= payload_type) {
return FALSE;
}
return TRUE;
}

char *janus_rtp_payload(char *buf, int len, int *plen) {
if(!buf || len < 12)
return NULL;
Expand Down
5 changes: 5 additions & 0 deletions src/rtp.h
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Expand Up @@ -154,6 +154,11 @@ int janus_videocodec_pt(janus_videocodec vcodec);
* @param[in] len Length of the buffer to inspect */
gboolean janus_is_rtp(char *buf, guint len);

/*! \brief Checks if the given payload type falls within the dynamic range [96, 127].
* @param[in] payload_type The payload type value to check.
* @returns TRUE if the payload type is in the dynamic range, FALSE otherwise. */
gboolean janus_is_dynamic_payload_type(int payload_type);

/*! \brief Helper to quickly access the RTP payload, skipping header and extensions
* @param[in] buf The packet data
* @param[in] len The packet data length in bytes
Expand Down