Skip to content

Commit

Permalink
Add support for "progress" request on SIP Plugin (#3466)
Browse files Browse the repository at this point in the history
  • Loading branch information
@adnan-mujagic
adnan-mujagic authored Nov 25, 2024
1 parent bcdaf4e commit b04234c
Showing 1 changed file with 83 additions and 35 deletions.
118 changes: 83 additions & 35 deletions src/plugins/janus_sip.c
View file
Original file line number Diff line number Diff line change
Expand Up @@ -36,16 +36,18 @@
* as events with the same transaction.
*
* The supported requests are \c register , \c unregister , \c call ,
* \c accept, \c decline , \c info , \c message , \c dtmf_info ,
* \progress , \c accept , \c decline , \c info , \c message , \c dtmf_info ,
* \c subscribe , \c unsubscribe , \c transfer , \c recording ,
* \c hold , \c unhold , \c update and \c hangup . \c register can be used,
* as the name suggests, to register a username at a SIP registrar to
* call and be called, while \c unregister unregisters it; \c call is used
* to send an INVITE to a different SIP URI through the plugin, while
* \c accept and \c decline are used to accept or reject the call in
* case one is invited instead of inviting; \c transfer takes care of
* attended and blind transfers (see \ref siptr for more details);
* \c hold and \c unhold can be used respectively to put a
* to send an INVITE to a different SIP URI through the plugin; in case one
* is invited instead of inviting, \c progress, \c accept and \c decline
* requests may be used. \c progress request is optional, and it is used to
* send 183 Session Progress response back to the caller, while
* \c accept and \c decline are used to accept or reject the call respectively;
* \c transfer takes care of attended and blind transfers (see \ref siptr for
* more details); \c hold and \c unhold can be used respectively to put a
* call on-hold and to resume it; \c info allows you to send a generic
* SIP INFO request, while \c dtmf_info is focused on using INFO for DTMF
* instead; \c message is the method you use to send a SIP message
Expand Down Expand Up @@ -299,8 +301,28 @@
\endverbatim
*
* The \c incomingcall may or may not be accompanied by a JSEP offer, depending
* on whether the caller sent an offerless INVITE or a regular one. Either
* way, you can accept the incoming call with the \c accept request:
* on whether the caller sent an offerless INVITE or a regular one. Optionally,
* you can progress the incoming call with the \c progress request:
*
\verbatim
{
"request" : "progress",
"srtp" : "<whether to mandate (sdes_mandatory) or offer (sdes_optional) SRTP support; optional>",
"headers" : "<object with key/value mappings (header name/value), to specify custom headers to add to the SIP OK; optional>"
"autoaccept_reinvites" : <true|false, whether we should blindly accept re-INVITEs with a 200 OK instead of relaying the SDP to the browser; optional, TRUE by default>
}
\endverbatim
*
* A \c progressing event will be sent back, as this is an asynchronous request.
*
* This will result in a <code>183 Session Progress</code> to be sent back to the caller.
* A \c progress request must always be accompanied by a JSEP answer (if the
* \c incomingcall event contained an offer) or offer (in case it was an
* offerless INVITE). This request can be used to inform the caller that the early
* media is available, such as ringback audio, announcements or other audio streams,
* without the call being fully established.
*
* Furthermore, you can accept the incoming call with the \c accept request:
*
\verbatim
{
Expand All @@ -314,13 +336,13 @@
* An \c accepting event will be sent back, as this is an asynchronous request.
*
* This will result in a <code>200 OK</code> to be sent back to the caller.
* An \c accept request must always be accompanied by a JSEP answer (if the
* \c incomingcall event contained an offer) or offer (in case it was an
* offerless INVITE). In the former case, an \c accepted event will be
* sent back just to confirm the call can be considered established;
* in the latter case, instead, an \c accepting event will be sent back
* instead, and an \c accepted event will only follow later, as soon as
* a JSEP answer is available in the SIP ACK the caller sent back.
* As was the case for \c progress request, an \c accept request must always
* be accompanied by a JSEP answer (if the \c incomingcall event contained an
* offer) or offer (in case it was an offerless INVITE). In the former case,
* an \c accepted event will be sent back just to confirm the call can be
* considered established; in the latter case, instead, an \c accepting event
* will be sent back instead, and an \c accepted event will only follow later,
* as soon as a JSEP answer is available in the SIP ACK the caller sent back.
*
* Notice that in case you get an incoming call while you're in another
* call, you will NOT get an \c incomingcall event, but a \c missed_call
Expand Down Expand Up @@ -808,6 +830,11 @@ static struct janus_json_parameter accept_parameters[] = {
{"headers", JSON_OBJECT, 0},
{"autoaccept_reinvites", JANUS_JSON_BOOL, 0}
};
static struct janus_json_parameter progress_parameters[] = {
{"srtp", JSON_STRING, 0},
{"headers", JSON_OBJECT, 0},
{"autoaccept_reinvites", JANUS_JSON_BOOL, 0}
};
static struct janus_json_parameter decline_parameters[] = {
{"code", JANUS_JSON_INTEGER, 0},
{"headers", JSON_OBJECT, 0},
Expand Down Expand Up @@ -920,6 +947,7 @@ typedef enum {
janus_sip_call_status_idle = 0,
janus_sip_call_status_inviting,
janus_sip_call_status_invited,
janus_sip_call_status_progress,
janus_sip_call_status_incall,
janus_sip_call_status_incall_reinviting,
janus_sip_call_status_incall_reinvited,
Expand All @@ -934,6 +962,8 @@ static const char *janus_sip_call_status_string(janus_sip_call_status status) {
return "inviting";
case janus_sip_call_status_invited:
return "invited";
case janus_sip_call_status_progress:
return "progress";
case janus_sip_call_status_incall:
return "incall";
case janus_sip_call_status_incall_reinviting:
Expand Down Expand Up @@ -2551,7 +2581,7 @@ void janus_sip_incoming_rtp(janus_plugin_session *handle, janus_plugin_rtp *pack
JANUS_LOG(LOG_ERR, "No session associated with this handle...\n");
return;
}
if(!janus_sip_call_is_established(session))
if(!janus_sip_call_is_established(session) && session->status != janus_sip_call_status_progress)
return;
gboolean video = packet->video;
char *buf = packet->buffer;
Expand Down Expand Up @@ -2679,7 +2709,7 @@ void janus_sip_incoming_rtcp(janus_plugin_session *handle, janus_plugin_rtcp *pa
JANUS_LOG(LOG_ERR, "No session associated with this handle...\n");
return;
}
if(!janus_sip_call_is_established(session))
if(!janus_sip_call_is_established(session) && session->status != janus_sip_call_status_progress)
return;
gboolean video = packet->video;
char *buf = packet->buffer;
Expand Down Expand Up @@ -3937,13 +3967,20 @@ static void *janus_sip_handler(void *data) {
result = json_object();
json_object_set_new(result, "event", json_string("calling"));
json_object_set_new(result, "call_id", json_string(session->callid));
} else if(!strcasecmp(request_text, "accept")) {
if(session->status != janus_sip_call_status_invited) {
} else if(!strcasecmp(request_text, "accept") || !strcasecmp(request_text, "progress")) {
gboolean progress = !strcasecmp(request_text, "progress");
if(progress && session->status != janus_sip_call_status_invited) {
JANUS_LOG(LOG_ERR, "Wrong state (not invited? status=%s)\n", janus_sip_call_status_string(session->status));
error_code = JANUS_SIP_ERROR_WRONG_STATE;
g_snprintf(error_cause, 512, "Wrong state (not invited? status=%s)", janus_sip_call_status_string(session->status));
goto error;
}
if(!progress && session->status != janus_sip_call_status_invited && session->status != janus_sip_call_status_progress) {
JANUS_LOG(LOG_ERR, "Wrong state (not invited or progress? status=%s)\n", janus_sip_call_status_string(session->status));
error_code = JANUS_SIP_ERROR_WRONG_STATE;
g_snprintf(error_cause, 512, "Wrong state (not invited or progress? status=%s)", janus_sip_call_status_string(session->status));
goto error;
}
janus_mutex_lock(&session->mutex);
if(session->callee == NULL) {
janus_mutex_unlock(&session->mutex);
Expand All @@ -3953,7 +3990,8 @@ static void *janus_sip_handler(void *data) {
goto error;
}
janus_mutex_unlock(&session->mutex);
JANUS_VALIDATE_JSON_OBJECT(root, accept_parameters,
struct janus_json_parameter *params = progress ? progress_parameters : accept_parameters;
JANUS_VALIDATE_JSON_OBJECT(root, params,
error_code, error_cause, TRUE,
JANUS_SIP_ERROR_MISSING_ELEMENT, JANUS_SIP_ERROR_INVALID_ELEMENT);
if(error_code != 0)
Expand Down Expand Up @@ -3982,9 +4020,9 @@ static void *janus_sip_handler(void *data) {
if(session->media.has_video)
has_srtp = (has_srtp && session->media.has_srtp_remote_video);
if(session->media.require_srtp && !has_srtp) {
JANUS_LOG(LOG_ERR, "Can't accept the call: SDES-SRTP required, but caller didn't offer it\n");
JANUS_LOG(LOG_ERR, "Can't %s the call: SDES-SRTP required, but caller didn't offer it\n", progress ? "progress" : "accept");
error_code = JANUS_SIP_ERROR_TOO_STRICT;
g_snprintf(error_cause, 512, "Can't accept the call: SDES-SRTP required, but caller didn't offer it");
g_snprintf(error_cause, 512, "Can't %s the call: SDES-SRTP required, but caller didn't offer it", progress ? "progress" : "accept");
goto error;
}
answer_srtp = answer_srtp || session->media.has_srtp_remote_audio || session->media.has_srtp_remote_video;
Expand All @@ -4006,8 +4044,8 @@ static void *janus_sip_handler(void *data) {
g_snprintf(error_cause, 512, "Media encryption unsupported by this plugin");
goto error;
}
/* Accept a call from another peer */
JANUS_LOG(LOG_VERB, "We're accepting the call from %s\n", session->callee);
/* Accept/Progress a call from another peer */
JANUS_LOG(LOG_VERB, "We're %s the call from %s\n", progress ? "progressing" : "accepting", session->callee);
gboolean answer = !strcasecmp(msg_sdp_type, "answer");
if(!answer) {
JANUS_LOG(LOG_VERB, "This is a response to an offerless INVITE\n");
Expand Down Expand Up @@ -4042,7 +4080,7 @@ static void *janus_sip_handler(void *data) {
session->media.has_video = TRUE; /* FIXME Maybe we need a better way to signal this */
}
janus_mutex_lock(&session->mutex);
if(janus_sip_allocate_local_ports(session, FALSE) < 0) {
if(janus_sip_allocate_local_ports(session, session->status == janus_sip_call_status_progress ? TRUE : FALSE) < 0) {
janus_mutex_unlock(&session->mutex);
JANUS_LOG(LOG_ERR, "Could not allocate RTP/RTCP ports\n");
janus_sdp_destroy(parsed_sdp);
Expand Down Expand Up @@ -4070,7 +4108,7 @@ static void *janus_sip_handler(void *data) {
/* Take note of the SDP (may be useful for UPDATEs or re-INVITEs) */
janus_sdp_destroy(session->sdp);
session->sdp = parsed_sdp;
JANUS_LOG(LOG_VERB, "Prepared SDP for 200 OK:\n%s", sdp);
JANUS_LOG(LOG_VERB, "Prepared SDP for %s:\n%s", progress ? "183 Session Progress" : "200 OK", sdp);
/* If the user negotiated simulcasting, just stick with the base substream */
json_t *msg_simulcast = json_object_get(msg->jsep, "simulcast");
if(msg_simulcast) {
Expand All @@ -4079,30 +4117,39 @@ static void *janus_sip_handler(void *data) {
if(s && json_array_size(s) > 0)
session->media.simulcast_ssrc = json_integer_value(json_array_get(s, 0));
}
const char *event_value;
if(progress) {
event_value = "progressed";
} else if(answer) {
event_value = "accepted";
} else {
event_value = "accepting";
}
/* Also notify event handlers */
if(notify_events && gateway->events_is_enabled()) {
json_t *info = json_object();
json_object_set_new(info, "event", json_string(answer ? "accepted" : "accepting"));
json_object_set_new(info, "event", json_string(event_value));
if(session->callid)
json_object_set_new(info, "call-id", json_string(session->callid));
gateway->notify_event(&janus_sip_plugin, session->handle, info);
}
/* Check if the OK needs to be enriched with custom headers */
char custom_headers[2048];
janus_sip_parse_custom_headers(root, (char *)&custom_headers, sizeof(custom_headers));
/* Send 200 OK */
/* Send 200 OK/183 Session progress */
if(!answer) {
if(session->transaction)
g_free(session->transaction);
session->transaction = msg->transaction ? g_strdup(msg->transaction) : NULL;
}
g_atomic_int_set(&session->hangingup, 0);
janus_sip_call_update_status(session, janus_sip_call_status_incall);
janus_sip_call_update_status(session, progress ? janus_sip_call_status_progress : janus_sip_call_status_incall);
if(session->stack->s_nh_i == NULL) {
JANUS_LOG(LOG_WARN, "NUA Handle for 200 OK still null??\n");
JANUS_LOG(LOG_WARN, "NUA Handle for %s null\n", progress ? "183 Session Progress" : "200 OK");
}
int sip_response = progress ? 183 : 200;
nua_respond(session->stack->s_nh_i,
200, sip_status_phrase(200),
sip_response, sip_status_phrase(sip_response),
SOATAG_USER_SDP_STR(sdp),
SOATAG_RTP_SELECT(SOA_RTP_SELECT_COMMON),
NUTAG_AUTOANSWER(0),
Expand All @@ -4112,7 +4159,7 @@ static void *janus_sip_handler(void *data) {
g_free(sdp);
/* Send an ack back */
result = json_object();
json_object_set_new(result, "event", json_string(answer ? "accepted" : "accepting"));
json_object_set_new(result, "event", json_string(event_value));
if(answer) {
/* Start the media */
session->media.ready = TRUE; /* FIXME Maybe we need a better way to signal this */
Expand Down Expand Up @@ -4349,7 +4396,7 @@ static void *janus_sip_handler(void *data) {
}
}
/* Reject an incoming call */
if(session->status != janus_sip_call_status_invited) {
if(session->status != janus_sip_call_status_invited && session->status != janus_sip_call_status_progress) {
JANUS_LOG(LOG_ERR, "Wrong state (not invited? status=%s)\n", janus_sip_call_status_string(session->status));
/* Ignore */
janus_sip_message_free(msg);
Expand Down Expand Up @@ -4593,8 +4640,8 @@ static void *janus_sip_handler(void *data) {
json_object_set_new(result, "event", json_string(hold ? "holding" : "resuming"));
} else if(!strcasecmp(request_text, "hangup")) {
/* Hangup an ongoing call */
if(!janus_sip_call_is_established(session) && session->status != janus_sip_call_status_inviting) {
JANUS_LOG(LOG_ERR, "Wrong state (not established/inviting? status=%s)\n",
if(!janus_sip_call_is_established(session) && session->status != janus_sip_call_status_inviting && session->status != janus_sip_call_status_progress) {
JANUS_LOG(LOG_ERR, "Wrong state (not established/inviting/progress? status=%s)\n",
janus_sip_call_status_string(session->status));
/* Ignore */
janus_sip_message_free(msg);
Expand Down Expand Up @@ -4630,6 +4677,7 @@ static void *janus_sip_handler(void *data) {
} else if(!strcasecmp(request_text, "recording")) {
/* Start or stop recording */
if(!(session->status == janus_sip_call_status_inviting || /* Presume it makes sense to start recording with early media? */
session->status == janus_sip_call_status_progress ||
janus_sip_call_is_established(session))) {
JANUS_LOG(LOG_ERR, "Wrong state (not in a call? status=%s)\n", janus_sip_call_status_string(session->status));
g_snprintf(error_cause, 512, "Wrong state (not in a call?)");
Expand Down

0 comments on commit b04234c

Please sign in to comment.