-
Caller
+
Caller
+
+
+
+
diff --git a/html/nosiptest.js b/html/nosiptest.js
index 2f1598b5cd..098e00ff14 100644
--- a/html/nosiptest.js
+++ b/html/nosiptest.js
@@ -56,6 +56,9 @@ var opaqueId = Janus.randomString(12);
var spinner = null;
+var videoenabled = true;
+var srtp = undefined ; // use "sdes_mandatory" to test SRTP-SDES
+
$(document).ready(function() {
// Initialize the library (all console debuggers enabled)
Janus.init({debug: "all", callback: function() {
@@ -91,7 +94,7 @@ $(document).ready(function() {
Janus.debug("[caller] Trying a createOffer too (audio/video sendrecv)");
caller.createOffer(
{
- // No media provided: by default, it's sendrecv for audio and video
+ media: {audio: true, video: videoenabled},
success: function(jsep) {
Janus.debug("[caller] Got SDP!", jsep);
// We now have a WebRTC SDP: to get a barebone SDP legacy
@@ -101,7 +104,8 @@ $(document).ready(function() {
// the SIP plugin uses (mandatory vs. optional). We'll
// get the result in an event called "generated" here.
var body = {
- request: "generate"
+ request: "generate",
+ srtp: srtp
};
caller.send({ message: body, jsep: jsep });
},
@@ -172,7 +176,8 @@ $(document).ready(function() {
request: "process",
type: result["type"],
sdp: result["sdp"],
- update: result["update"]
+ update: result["update"],
+ srtp: srtp
}
callee.send({ message: processOffer });
} else if(event === "processed") {
@@ -273,6 +278,43 @@ $(document).ready(function() {
$('#videoright .no-video-container').remove();
$('#peervideo').removeClass('hide').show();
}
+
+ if(videoenabled) {
+ $('#togglevideo').html("Disable video").removeClass("btn-success").addClass("btn-danger");
+ } else {
+ $('#togglevideo').html("Enable video").removeClass("btn-danger").addClass("btn-success");
+ }
+
+ $('#togglevideo').unbind('click').removeAttr('disabled').click(
+ function() {
+ videoenabled = !videoenabled;
+ var media;
+ if(videoenabled) {
+ $('#togglevideo').html("Disable video").removeClass("btn-success").addClass("btn-danger");
+ media = {addVideo: true};
+ } else {
+ $('#togglevideo').html("Enable video").removeClass("btn-danger").addClass("btn-success");
+ media = {removeVideo: true};
+ }
+ caller.createOffer(
+ {
+ media: media,
+ success: function(jsep) {
+ Janus.debug("[caller] Got UPDATE SDP!");
+ Janus.debug(jsep);
+ var body = {
+ request: "generate",
+ update: true,
+ srtp: srtp
+ };
+ caller.send({message: body, jsep: jsep});
+ },
+ error: function(error) {
+ Janus.error("WebRTC error:", error);
+ bootbox.alert("WebRTC error... " + JSON.stringify(error));
+ }
+ });
+ });
},
oncleanup: function() {
Janus.log("[caller] ::: Got a cleanup notification :::");
@@ -363,7 +405,8 @@ $(document).ready(function() {
// We'll get the result in an event called "generated" here.
var body = {
request: "generate",
- update: update
+ update: update,
+ srtp: srtp
};
callee.send({ message: body, jsep: jsep });
},
@@ -385,7 +428,8 @@ $(document).ready(function() {
request: "process",
type: result["type"],
sdp: result["sdp"],
- update: result["update"]
+ update: result["update"],
+ srtp: srtp
}
caller.send({ message: processAnswer });
}
diff --git a/plugins/janus_nosip.c b/plugins/janus_nosip.c
index 87d8ab524b..df8b3dd540 100644
--- a/plugins/janus_nosip.c
+++ b/plugins/janus_nosip.c
@@ -300,6 +300,7 @@ typedef struct janus_nosip_media {
const char *audio_pt_name;
srtp_t audio_srtp_in, audio_srtp_out;
srtp_policy_t audio_remote_policy, audio_local_policy;
+ char *audio_srtp_local_profile, *audio_srtp_local_crypto;
gboolean audio_send;
int has_video:1;
int video_rtp_fd, video_rtcp_fd;
@@ -311,6 +312,7 @@ typedef struct janus_nosip_media {
const char *video_pt_name;
srtp_t video_srtp_in, video_srtp_out;
srtp_policy_t video_remote_policy, video_local_policy;
+ char *video_srtp_local_profile, *video_srtp_local_crypto;
gboolean video_send;
janus_rtp_switching_context context;
int pipefd[2];
@@ -575,6 +577,14 @@ static void janus_nosip_srtp_cleanup(janus_nosip_session *session) {
session->media.audio_srtp_in = NULL;
g_free(session->media.audio_remote_policy.key);
session->media.audio_remote_policy.key = NULL;
+ if(session->media.audio_srtp_local_profile) {
+ g_free(session->media.audio_srtp_local_profile);
+ session->media.audio_srtp_local_profile = NULL;
+ }
+ if(session->media.audio_srtp_local_crypto) {
+ g_free(session->media.audio_srtp_local_crypto);
+ session->media.audio_srtp_local_crypto = NULL;
+ }
/* Video */
if(session->media.video_srtp_out)
srtp_dealloc(session->media.video_srtp_out);
@@ -586,6 +596,14 @@ static void janus_nosip_srtp_cleanup(janus_nosip_session *session) {
session->media.video_srtp_in = NULL;
g_free(session->media.video_remote_policy.key);
session->media.video_remote_policy.key = NULL;
+ if(session->media.video_srtp_local_profile) {
+ g_free(session->media.video_srtp_local_profile);
+ session->media.video_srtp_local_profile = NULL;
+ }
+ if(session->media.video_srtp_local_crypto) {
+ g_free(session->media.video_srtp_local_crypto);
+ session->media.video_srtp_local_crypto = NULL;
+ }
}
void janus_nosip_media_reset(janus_nosip_session *session) {
@@ -868,6 +886,10 @@ void janus_nosip_create_session(janus_plugin_session *handle, int *error) {
session->media.has_srtp_local = FALSE;
session->media.has_srtp_remote = FALSE;
session->media.srtp_profile = 0;
+ session->media.audio_srtp_local_profile = NULL;
+ session->media.audio_srtp_local_crypto = NULL;
+ session->media.video_srtp_local_profile = NULL;
+ session->media.video_srtp_local_crypto = NULL;
session->media.has_audio = 0;
session->media.audio_rtp_fd = -1;
session->media.audio_rtcp_fd = -1;
@@ -1331,68 +1353,66 @@ static void *janus_nosip_handler(void *data) {
const char *info = json_string_value(json_object_get(root, "info"));
/* SDES-SRTP is disabled by default, let's see if we need to enable it */
gboolean do_srtp = FALSE, require_srtp = FALSE;
- janus_srtp_profile srtp_profile = JANUS_SRTP_AES128_CM_SHA1_80;
+ json_t *srtp = json_object_get(root, "srtp");
+ if(srtp) {
+ const char *srtp_text = json_string_value(srtp);
+ if(!strcasecmp(srtp_text, "sdes_optional")) {
+ /* Negotiate SDES, but make it optional */
+ do_srtp = TRUE;
+ } else if(!strcasecmp(srtp_text, "sdes_mandatory")) {
+ /* Negotiate SDES, and require it */
+ do_srtp = TRUE;
+ require_srtp = TRUE;
+ } else {
+ JANUS_LOG(LOG_ERR, "Invalid element (srtp can only be sdes_optional or sdes_mandatory)\n");
+ error_code = JANUS_NOSIP_ERROR_INVALID_ELEMENT;
+ g_snprintf(error_cause, 512, "Invalid element (srtp can only be sdes_optional or sdes_mandatory)");
+ goto error;
+ }
+ }
+ if(offer && !sdp_update) {
+ /* Clean up SRTP stuff from before first, in case it's still needed */
+ janus_nosip_srtp_cleanup(session);
+ if(do_srtp) {
+ JANUS_LOG(LOG_VERB, "Going to negotiate SDES-SRTP (%s)...\n", require_srtp ? "mandatory" : "optional");
+ }
+ }
+ session->media.require_srtp = require_srtp;
if(generate) {
- json_t *srtp = json_object_get(root, "srtp");
- if(srtp) {
- const char *srtp_text = json_string_value(srtp);
- if(!strcasecmp(srtp_text, "sdes_optional")) {
- /* Negotiate SDES, but make it optional */
- do_srtp = TRUE;
- } else if(!strcasecmp(srtp_text, "sdes_mandatory")) {
- /* Negotiate SDES, and require it */
- do_srtp = TRUE;
- require_srtp = TRUE;
- } else {
- JANUS_LOG(LOG_ERR, "Invalid element (srtp can only be sdes_optional or sdes_mandatory)\n");
- error_code = JANUS_NOSIP_ERROR_INVALID_ELEMENT;
- g_snprintf(error_cause, 512, "Invalid element (srtp can only be sdes_optional or sdes_mandatory)");
+ if(!offer) {
+ do_srtp = do_srtp || session->media.has_srtp_remote;
+ /* Make sure the request is consistent with the state (original offer) */
+ if(session->media.require_srtp && !session->media.has_srtp_remote) {
+ JANUS_LOG(LOG_ERR, "Can't generate answer: SDES-SRTP required, but caller didn't offer it\n");
+ error_code = JANUS_NOSIP_ERROR_TOO_STRICT;
+ g_snprintf(error_cause, 512, "Can't generate answer: SDES-SRTP required, but caller didn't offer it");
goto error;
}
- if(do_srtp) {
- /* Any SRTP profile different from the default? */
- srtp_profile = JANUS_SRTP_AES128_CM_SHA1_80;
- const char *profile = json_string_value(json_object_get(root, "srtp_profile"));
- if(profile) {
- if(!strcmp(profile, "AES_CM_128_HMAC_SHA1_32")) {
- srtp_profile = JANUS_SRTP_AES128_CM_SHA1_32;
- } else if(!strcmp(profile, "AES_CM_128_HMAC_SHA1_80")) {
- srtp_profile = JANUS_SRTP_AES128_CM_SHA1_80;
+ }
+ session->media.has_srtp_local = do_srtp;
+ if(do_srtp) {
+ /* Any SRTP profile different from the default? */
+ janus_srtp_profile srtp_profile = JANUS_SRTP_AES128_CM_SHA1_80;
+ const char *profile = json_string_value(json_object_get(root, "srtp_profile"));
+ if(profile) {
+ if(!strcmp(profile, "AES_CM_128_HMAC_SHA1_32")) {
+ srtp_profile = JANUS_SRTP_AES128_CM_SHA1_32;
+ } else if(!strcmp(profile, "AES_CM_128_HMAC_SHA1_80")) {
+ srtp_profile = JANUS_SRTP_AES128_CM_SHA1_80;
#ifdef HAVE_SRTP_AESGCM
} else if(!strcmp(profile, "AEAD_AES_128_GCM")) {
srtp_profile = JANUS_SRTP_AEAD_AES_128_GCM;
} else if(!strcmp(profile, "AEAD_AES_256_GCM")) {
srtp_profile = JANUS_SRTP_AEAD_AES_256_GCM;
#endif
- } else {
- JANUS_LOG(LOG_ERR, "Invalid element (unsupported SRTP profile)\n");
- error_code = JANUS_NOSIP_ERROR_INVALID_ELEMENT;
- g_snprintf(error_cause, 512, "Invalid element (unsupported SRTP profile)");
- goto error;
- }
- }
- }
- }
- if(!sdp_update) {
- if(offer) {
- /* Clean up SRTP stuff from before first, in case it's still needed */
- janus_nosip_srtp_cleanup(session);
- session->media.require_srtp = require_srtp;
- session->media.has_srtp_local = do_srtp;
- session->media.srtp_profile = srtp_profile;
- if(do_srtp) {
- JANUS_LOG(LOG_VERB, "Going to negotiate SDES-SRTP (%s)...\n", require_srtp ? "mandatory" : "optional");
- }
- } else {
- /* Make sure the request is consistent with the state (original offer) */
- if(session->media.require_srtp && !session->media.has_srtp_remote) {
- JANUS_LOG(LOG_ERR, "Can't generate answer: SDES-SRTP required, but caller didn't offer it\n");
- error_code = JANUS_NOSIP_ERROR_TOO_STRICT;
- g_snprintf(error_cause, 512, "Can't generate answer: SDES-SRTP required, but caller didn't offer it");
+ } else {
+ JANUS_LOG(LOG_ERR, "Invalid element (unsupported SRTP profile)\n");
+ error_code = JANUS_NOSIP_ERROR_INVALID_ELEMENT;
+ g_snprintf(error_cause, 512, "Invalid element (unsupported SRTP profile)");
goto error;
}
- do_srtp = do_srtp || session->media.has_srtp_remote;
}
+ session->media.srtp_profile = srtp_profile;
}
}
/* Get video-orientation extension id from SDP we got */
@@ -1466,10 +1486,6 @@ static void *janus_nosip_handler(void *data) {
} else {
/* We got a barebone offer or answer from our peer: process it accordingly */
gboolean changed = FALSE;
- if(!sdp_update && offer) {
- /* Clean up SRTP stuff from before first, in case it's still needed */
- janus_nosip_srtp_cleanup(session);
- }
janus_nosip_sdp_process(session, parsed_sdp, !offer, sdp_update, &changed);
/* Check if offer has neither audio nor video, fail */
if(!session->media.has_audio && !session->media.has_video) {
@@ -1487,6 +1503,12 @@ static void *janus_nosip_handler(void *data) {
g_snprintf(error_cause, 512, "No remote IP addresses");
goto error;
}
+ if(session->media.require_srtp && !session->media.has_srtp_remote) {
+ JANUS_LOG(LOG_ERR, "Can't process request: SDES-SRTP required, but caller didn't offer it\n");
+ error_code = JANUS_NOSIP_ERROR_TOO_STRICT;
+ g_snprintf(error_cause, 512, "Can't process request: SDES-SRTP required, but caller didn't offer it");
+ goto error;
+ }
/* Take note of the SDP (may be useful for UPDATEs or re-INVITEs) */
janus_sdp_destroy(session->sdp);
session->sdp = parsed_sdp;
@@ -1800,17 +1822,17 @@ void janus_nosip_sdp_process(janus_nosip_session *session, janus_sdp *sdp, gbool
g_free(session->media.remote_video_ip);
session->media.remote_video_ip = g_strdup(m->c_addr);
}
- if(update) {
- /* FIXME This is a session update, we only accept changes in IP/ports */
- temp = temp->next;
- continue;
- }
GList *tempA = m->attributes;
while(tempA) {
janus_sdp_attribute *a = (janus_sdp_attribute *)tempA->data;
if(a->name) {
if(!strcasecmp(a->name, "crypto")) {
if(m->type == JANUS_SDP_AUDIO || m->type == JANUS_SDP_VIDEO) {
+ if((m->type == JANUS_SDP_AUDIO && session->media.audio_srtp_in != NULL) || (m->type == JANUS_SDP_VIDEO && session->media.video_srtp_in != NULL)) {
+ /* Remote SRTP is already set */
+ tempA = tempA->next;
+ continue;
+ }
gint32 tag = 0;
char profile[101], crypto[101];
/* FIXME inline can be more complex than that, and we're currently only offering SHA1_80 */
@@ -1873,23 +1895,19 @@ char *janus_nosip_sdp_manipulate(janus_nosip_session *session, janus_sdp *sdp, g
if(m->type == JANUS_SDP_AUDIO) {
m->port = session->media.local_audio_rtp_port;
if(session->media.has_srtp_local) {
- char *profile = NULL;
- char *crypto = NULL;
- janus_nosip_srtp_set_local(session, FALSE, &profile, &crypto);
- janus_sdp_attribute *a = janus_sdp_attribute_create("crypto", "1 %s inline:%s", profile, crypto);
- g_free(profile);
- g_free(crypto);
+ if(!session->media.audio_srtp_local_profile || !session->media.audio_srtp_local_crypto) {
+ janus_nosip_srtp_set_local(session, FALSE, &session->media.audio_srtp_local_profile, &session->media.audio_srtp_local_crypto);
+ }
+ janus_sdp_attribute *a = janus_sdp_attribute_create("crypto", "1 %s inline:%s", session->media.audio_srtp_local_profile, session->media.audio_srtp_local_crypto);
m->attributes = g_list_append(m->attributes, a);
}
} else if(m->type == JANUS_SDP_VIDEO) {
m->port = session->media.local_video_rtp_port;
if(session->media.has_srtp_local) {
- char *profile = NULL;
- char *crypto = NULL;
- janus_nosip_srtp_set_local(session, TRUE, &profile, &crypto);
- janus_sdp_attribute *a = janus_sdp_attribute_create("crypto", "1 %s inline:%s", profile, crypto);
- g_free(profile);
- g_free(crypto);
+ if(!session->media.video_srtp_local_profile || !session->media.video_srtp_local_crypto) {
+ janus_nosip_srtp_set_local(session, TRUE, &session->media.video_srtp_local_profile, &session->media.video_srtp_local_crypto);
+ }
+ janus_sdp_attribute *a = janus_sdp_attribute_create("crypto", "1 %s inline:%s", session->media.video_srtp_local_profile, session->media.video_srtp_local_crypto);
m->attributes = g_list_append(m->attributes, a);
}
}
@@ -2208,8 +2226,6 @@ static void *janus_nosip_relay_thread(void *data) {
/* Loop */
int num = 0;
gboolean goon = TRUE;
- int astep = 0, vstep = 0;
- guint32 ats = 0, vts = 0;
session->media.updated = TRUE; /* Connect UDP sockets upon loop entry */
gboolean have_audio_server_ip = TRUE;
@@ -2409,25 +2425,6 @@ static void *janus_nosip_relay_thread(void *data) {
/* Check if the SSRC changed (e.g., after a re-INVITE or UPDATE) */
guint32 timestamp = ntohl(header->timestamp);
janus_rtp_header_update(header, &session->media.context, video, 0);
- if(video) {
- if(vts == 0) {
- vts = timestamp;
- } else if(vstep == 0) {
- vstep = timestamp-vts;
- if(vstep < 0) {
- vstep = 0;
- }
- }
- } else {
- if(ats == 0) {
- ats = timestamp;
- } else if(astep == 0) {
- astep = timestamp-ats;
- if(astep < 0) {
- astep = 0;
- }
- }
- }
/* Save the frame if we're recording */
janus_recorder_save_frame(video ? session->vrc_peer : session->arc_peer, buffer, bytes);
/* Relay to browser */