[Pkg-voip-commits] [janus] 82/282: Added support for offerless INVITEs to the SIPre plugin
Jonas Smedegaard
dr at jones.dk
Wed Dec 20 21:53:30 UTC 2017
This is an automated email from the git hooks/post-receive script.
js pushed a commit to annotated tag debian/0.2.6-1
in repository janus.
commit 486455a0309a13fa92e429f2f0ce49db6ab57a81
Author: Lorenzo Miniero <lminiero at gmail.com>
Date: Tue Jun 27 16:29:54 2017 +0200
Added support for offerless INVITEs to the SIPre plugin
---
html/sipretest.js | 21 ++++++--
plugins/janus_sipre.c | 145 ++++++++++++++++++++++++++++++++------------------
2 files changed, 109 insertions(+), 57 deletions(-)
diff --git a/html/sipretest.js b/html/sipretest.js
index 24f8292..256abab 100644
--- a/html/sipretest.js
+++ b/html/sipretest.js
@@ -206,12 +206,18 @@ $(document).ready(function() {
} else if(event === 'incomingcall') {
Janus.log("Incoming call from " + result["displayname"] + " (" + result["username"] + ")!");
var doAudio = true, doVideo = true;
+ var offerlessInvite = false;
if(jsep !== null && jsep !== undefined) {
// What has been negotiated?
doAudio = (jsep.sdp.indexOf("m=audio ") > -1);
doVideo = (jsep.sdp.indexOf("m=video ") > -1);
Janus.debug("Audio " + (doAudio ? "has" : "has NOT") + " been negotiated");
Janus.debug("Video " + (doVideo ? "has" : "has NOT") + " been negotiated");
+ } else {
+ Janus.log("This call doesn't contain an offer... we'll need to provide one ourselves");
+ offerlessInvite = true;
+ // In case you want to offer video when reacting to an offerless call, set this to true
+ doVideo = false;
}
// Any security offered? A missing "srtp" attribute means plain RTP
var rtpType = "";
@@ -222,8 +228,11 @@ $(document).ready(function() {
rtpType = " (SDES-SRTP mandatory)";
// Notify user
bootbox.hideAll();
+ var extra = "";
+ if(offerlessInvite)
+ extra = " (no SDP offer provided)";
incoming = bootbox.dialog({
- message: "Incoming call from " + result["displayname"] + " (" + result["username"] + ")!" + rtpType,
+ message: "Incoming call from " + result["displayname"] + " (" + result["username"] + ")!" + rtpType + extra,
title: "Incoming call",
closeButton: false,
buttons: {
@@ -233,12 +242,15 @@ $(document).ready(function() {
callback: function() {
incoming = null;
$('#peer').val(result["username"]).attr('disabled', true);
- sipcall.createAnswer(
+ // Notice that we can only answer if we got an offer: if this was
+ // an offerless call, we'll need to create an offer ourselves
+ var sipcallAction = (offerlessInvite ? sipcall.createOffer : sipcall.createAnswer);
+ sipcallAction(
{
jsep: jsep,
media: { audio: doAudio, video: doVideo },
success: function(jsep) {
- Janus.debug("Got SDP! audio=" + doAudio + ", video=" + doVideo);
+ Janus.debug("Got SDP " + jsep.type + "! audio=" + doAudio + ", video=" + doVideo);
Janus.debug(jsep);
var body = { request: "accept" };
// Note: as with "call", you can add a "srtp" attribute to
@@ -278,7 +290,8 @@ $(document).ready(function() {
});
} else if(event === 'accepted') {
Janus.log(result["username"] + " accepted the call!");
- // TODO Video call can start
+ Janus.log(jsep);
+ // Call can start, now: handle the remote answer
if(jsep !== null && jsep !== undefined) {
sipcall.handleRemoteJsep({jsep: jsep, error: doHangup });
}
diff --git a/plugins/janus_sipre.c b/plugins/janus_sipre.c
index 3966181..7e4a57d 100644
--- a/plugins/janus_sipre.c
+++ b/plugins/janus_sipre.c
@@ -1253,8 +1253,8 @@ void janus_sipre_incoming_rtp(janus_plugin_session *handle, int video, char *buf
video ? "video" : "audio",
video ? session->media.video_ssrc : session->media.audio_ssrc);
}
- if((video && session->media.has_video && session->media.video_rtp_fd) ||
- (!video && session->media.has_audio && session->media.audio_rtp_fd)) {
+ if((video && session->media.has_video && session->media.video_rtp_fd != -1) ||
+ (!video && session->media.has_audio && session->media.audio_rtp_fd != -1)) {
/* Save the frame if we're recording */
janus_recorder_save_frame(video ? session->vrc : session->arc, buf, len);
/* Is SRTP involved? */
@@ -1294,8 +1294,8 @@ void janus_sipre_incoming_rtcp(janus_plugin_session *handle, int video, char *bu
return;
}
/* Forward to our SIPre peer */
- if((video && session->media.has_video && session->media.video_rtcp_fd) ||
- (!video && session->media.has_audio && session->media.audio_rtcp_fd)) {
+ if((video && session->media.has_video && session->media.video_rtcp_fd != -1) ||
+ (!video && session->media.has_audio && session->media.audio_rtcp_fd != -1)) {
/* Fix SSRCs as the gateway does */
JANUS_LOG(LOG_HUGE, "[SIPre-%s] Fixing %s SSRCs (local %u, peer %u)\n",
session->account.username ? session->account.username : "unknown",
@@ -1965,6 +1965,10 @@ static void *janus_sipre_handler(void *data) {
}
/* Accept a call from another peer */
JANUS_LOG(LOG_VERB, "We're accepting the call from %s\n", session->callee);
+ gboolean answer = !strcasecmp(msg_sdp_type, "answer");
+ if(!answer) {
+ JANUS_LOG(LOG_VERB, "This is a response to an offerless INVITE\n");
+ }
JANUS_LOG(LOG_VERB, "This is involving a negotiation (%s) as well:\n%s\n", msg_sdp_type, msg_sdp);
session->media.has_srtp_local = answer_srtp;
if(answer_srtp) {
@@ -2018,27 +2022,34 @@ static void *janus_sipre_handler(void *data) {
/* Also notify event handlers */
if(notify_events && gateway->events_is_enabled()) {
json_t *info = json_object();
- json_object_set_new(info, "event", json_string("accepted"));
+ json_object_set_new(info, "event", json_string(answer ? "accepted" : "accepting"));
if(session->callid)
json_object_set_new(info, "call-id", json_string(session->callid));
gateway->notify_event(&janus_sipre_plugin, session->handle, info);
}
/* Enqueue the 200 OK */
+ 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);
session->status = janus_sipre_call_status_incall;
session->temp_sdp = sdp;
mqueue_push(mq, janus_sipre_mqueue_event_do_accept, janus_sipre_mqueue_payload_create(session, NULL, 0, NULL));
/* Send an ack back */
result = json_object();
- json_object_set_new(result, "event", json_string("accepted"));
- /* Start the media */
- session->media.ready = TRUE; /* FIXME Maybe we need a better way to signal this */
- GError *error = NULL;
- char tname[16];
- g_snprintf(tname, sizeof(tname), "siprertp %s", session->account.username);
- g_thread_try_new(tname, janus_sipre_relay_thread, session, &error);
- if(error != NULL) {
- JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the RTP/RTCP thread...\n", error->code, error->message ? error->message : "??");
+ json_object_set_new(result, "event", json_string(answer ? "accepted" : "accepting"));
+ if(answer) {
+ /* Start the media */
+ session->media.ready = TRUE; /* FIXME Maybe we need a better way to signal this */
+ GError *error = NULL;
+ char tname[16];
+ g_snprintf(tname, sizeof(tname), "siprertp %s", session->account.username);
+ g_thread_try_new(tname, janus_sipre_relay_thread, session, &error);
+ if(error != NULL) {
+ JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the RTP/RTCP thread...\n", error->code, error->message ? error->message : "??");
+ }
}
} else if(!strcasecmp(request_text, "decline")) {
/* Reject an incoming call */
@@ -2927,13 +2938,37 @@ static void *janus_sipre_relay_thread(void *data) {
int i = 0;
for(i=0; i<num; i++) {
if(fds[i].revents & (POLLERR | POLLHUP)) {
- /* Socket error? */
- JANUS_LOG(LOG_ERR, "[SIPre-%s] Error polling: %s...\n", session->account.username,
- fds[i].revents & POLLERR ? "POLLERR" : "POLLHUP");
- JANUS_LOG(LOG_ERR, "[SIPre-%s] -- %d (%s)\n", session->account.username, errno, strerror(errno));
+ /* If we just updated the session, let's wait until things have calmed down */
if(session->media.updated)
break;
- goon = FALSE; /* Can we assume it's pretty much over, after a POLLERR? */
+ /* Check the socket error */
+ int error = 0;
+ socklen_t errlen = sizeof(error);
+ getsockopt(fds[i].fd, SOL_SOCKET, SO_ERROR, (void *)&error, &errlen);
+ if(error == 0) {
+ /* Maybe not a breaking error after all? */
+ continue;
+ } else if(error == 111) {
+ /* ICMP error? If it's related to RTCP, let's just close the RTCP socket and move on */
+ if(fds[i].fd == session->media.audio_rtcp_fd) {
+ JANUS_LOG(LOG_WARN, "[SIPre-%s] Got a '%s' on the audio RTCP socket, closing it\n",
+ session->account.username, strerror(error));
+ close(session->media.audio_rtcp_fd);
+ session->media.audio_rtcp_fd = -1;
+ } else if(fds[i].fd == session->media.video_rtcp_fd) {
+ JANUS_LOG(LOG_WARN, "[SIPre-%s] Got a '%s' on the video RTCP socket, closing it\n",
+ session->account.username, strerror(error));
+ close(session->media.video_rtcp_fd);
+ session->media.video_rtcp_fd = -1;
+ }
+ /* FIXME Should we do the same with the RTP sockets as well? We may risk overreacting, there... */
+ continue;
+ }
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] Error polling %d (socket #%d): %s...\n", session->account.username,
+ fds[i].fd, i, fds[i].revents & POLLERR ? "POLLERR" : "POLLHUP");
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] -- %d (%s)\n", session->account.username, error, strerror(error));
+ /* Can we assume it's pretty much over, after a POLLERR? */
+ goon = FALSE;
/* FIXME Simulate a "hangup" coming from the browser */
janus_sipre_message *msg = g_malloc0(sizeof(janus_sipre_message));
msg->handle = session->handle;
@@ -3263,24 +3298,23 @@ void janus_sipre_cb_incoming(const struct sip_msg *msg, void *arg) {
}
return;
}
- /* New incoming call */
- const char *offer = (const char *)mbuf_buf(msg->mb);
- if(offer == NULL) {
- /* No SDP? */
- JANUS_LOG(LOG_WARN, "[SIPre-%s] No SDP in the INVITE?\n", session->account.username);
- mqueue_push(mq, janus_sipre_mqueue_event_do_rcode, janus_sipre_mqueue_payload_create(session, msg, 488, NULL));
- return;
- }
+ /* New incoming call, check if there's an SDP to process */
char sdp_offer[1024];
- g_snprintf(sdp_offer, sizeof(sdp_offer), "%.*s", (int)mbuf_get_left(msg->mb), offer);
- JANUS_LOG(LOG_HUGE, "[SIPre-%s] -- Offer: %s\n", session->account.username, sdp_offer);
- /* Parse the remote SDP */
- char sdperror[100];
- janus_sdp *sdp = janus_sdp_parse(sdp_offer, sdperror, sizeof(sdperror));
- if(!sdp) {
- JANUS_LOG(LOG_ERR, "Error parsing SDP! %s\n", sdperror);
- mqueue_push(mq, janus_sipre_mqueue_event_do_rcode, janus_sipre_mqueue_payload_create(session, msg, 488, NULL));
- return;
+ janus_sdp *sdp = NULL;
+ const char *offer = (const char *)mbuf_buf(msg->mb);
+ if(offer == NULL || mbuf_get_left(msg->mb) == 0) {
+ JANUS_LOG(LOG_WARN, "[SIPre-%s] Received offerless INVITE\n", session->account.username);
+ } else {
+ g_snprintf(sdp_offer, sizeof(sdp_offer), "%.*s", (int)mbuf_get_left(msg->mb), offer);
+ JANUS_LOG(LOG_WARN, "[SIPre-%s] -- Offer: %s\n", session->account.username, sdp_offer);
+ /* Parse the remote SDP */
+ char sdperror[100];
+ sdp = janus_sdp_parse(sdp_offer, sdperror, sizeof(sdperror));
+ if(!sdp) {
+ JANUS_LOG(LOG_ERR, "Error parsing SDP! %s\n", sdperror);
+ mqueue_push(mq, janus_sipre_mqueue_event_do_rcode, janus_sipre_mqueue_payload_create(session, msg, 488, NULL));
+ return;
+ }
}
session->callee = g_strdup(from);
session->callid = g_strdup(callid);
@@ -3290,23 +3324,27 @@ void janus_sipre_cb_incoming(const struct sip_msg *msg, void *arg) {
janus_sipre_srtp_cleanup(session);
/* Parse SDP */
JANUS_LOG(LOG_VERB, "Someone is inviting us a call\n");
- gboolean changed = FALSE;
- janus_sipre_sdp_process(session, sdp, FALSE, FALSE, &changed);
- /* Check if offer has neither audio nor video, fail with 488 */
- if (!session->media.has_audio && !session->media.has_video) {
- mqueue_push(mq, janus_sipre_mqueue_event_do_rcode, janus_sipre_mqueue_payload_create(session, msg, 488, NULL));
- janus_sdp_free(sdp);
- return;
- }
- /* Also fail with 488 if there's no remote IP address that can be used for RTP */
- if (!session->media.remote_ip) {
- mqueue_push(mq, janus_sipre_mqueue_event_do_rcode, janus_sipre_mqueue_payload_create(session, msg, 488, NULL));
- janus_sdp_free(sdp);
- return;
+ if(sdp) {
+ gboolean changed = FALSE;
+ janus_sipre_sdp_process(session, sdp, FALSE, FALSE, &changed);
+ /* Check if offer has neither audio nor video, fail with 488 */
+ if(!session->media.has_audio && !session->media.has_video) {
+ mqueue_push(mq, janus_sipre_mqueue_event_do_rcode, janus_sipre_mqueue_payload_create(session, msg, 488, NULL));
+ janus_sdp_free(sdp);
+ return;
+ }
+ /* Also fail with 488 if there's no remote IP address that can be used for RTP */
+ if(!session->media.remote_ip) {
+ mqueue_push(mq, janus_sipre_mqueue_event_do_rcode, janus_sipre_mqueue_payload_create(session, msg, 488, NULL));
+ janus_sdp_free(sdp);
+ return;
+ }
}
session->stack.invite = msg;
- /* Send SDP to the browser */
- json_t *jsep = json_pack("{ssss}", "type", "offer", "sdp", sdp_offer);
+ /* Notify the browser about the call */
+ json_t *jsep = NULL;
+ if(sdp)
+ jsep = json_pack("{ssss}", "type", "offer", "sdp", sdp_offer);
json_t *call = json_object();
json_object_set_new(call, "sip", json_string("event"));
json_t *calling = json_object();
@@ -3315,7 +3353,7 @@ void janus_sipre_cb_incoming(const struct sip_msg *msg, void *arg) {
if(strlen(dname)) {
json_object_set_new(calling, "displayname", json_string(dname));
}
- if(session->media.has_srtp_remote) {
+ if(sdp && session->media.has_srtp_remote) {
/* FIXME Maybe a true/false instead? */
json_object_set_new(calling, "srtp", json_string(session->media.require_srtp ? "sdes_mandatory" : "sdes_optional"));
}
@@ -3325,7 +3363,8 @@ void janus_sipre_cb_incoming(const struct sip_msg *msg, void *arg) {
JANUS_LOG(LOG_VERB, " >> Pushing event to peer: %d (%s)\n", ret, janus_get_api_error(ret));
}
json_decref(call);
- json_decref(jsep);
+ if(jsep)
+ json_decref(jsep);
janus_sdp_free(sdp);
/* Also notify event handlers */
if(notify_events && gateway->events_is_enabled()) {
--
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