search for: wss

...y distance matrix (note: I performed k-means on the MDS coordinates because applying a euclidean distance measure to my raw data would have been inappropriate) canto.MDS<-cmdscale(canto) I then figured out what would be my optimum k-value by plotting the within sums of squares for K1-K15 > wss <- (nrow(canto.MDS)-1)*sum(apply(canto.MDS,2,var)) > wss[2] <- sum(kmeans(canto.MDS,centers=2)$withinss) > wss[3] <- sum(kmeans(canto.MDS,centers=3)$withinss) > wss[4] <- sum(kmeans(canto.MDS,centers=4)$withinss) > wss[5] <- sum(kmeans(canto.MDS,centers=5)$withinss)...

Hey all. I have webrtc up and running with asterisk 11. All is going well with TLS now working. At least I hope it is using TLS and wss. Based on what I am seeing I have UDP, WSS listed in the Allowed transports, but every time I connect the Primary transport shows WS.. Why is this? Am I actually running ws in wss mode? Prim.Transp. : WS Allowed.Trsp : UDP,WSS Def. Username: 6167761066.2011 SIP Options : (none)...

Hi, Have anyone tried using SIPML5 to connect to Asterisk over wss? I'm having the error as shown below Connecting to 'wss://54.xxx.xxx.xxx:8080/ws <wss://54.254.228.251:8080/ws>' SIPml-api.js?svn=224:1 ==stack event = starting SIPml-api.js?svn=224:1 __tsip_transport_ws_onerror SIPml-api.js?svn=224:1 __tsip_transport_ws_onclose SIPml-api.js?svn...

...key=/etc/asterisk/keys/mycert.pem ; path to private key file (*.pem) only. Date: Tue, 13 Jan 2015 10:02:08 +0000 From: Alexej Starschenko <a.starschenko at sabienzia.com> To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> Subject: [asterisk-users] WSS Socket Configuration Message-ID: <46B08B12850FEF43BF479468A4002A57829D17 at SZ-ORG-APP001.sabienzia.int> Content-Type: text/plain; charset="us-ascii" Hi, I have a working WebRTC/SipJS+Asterisk(13.0.1) setup using ws sockets. Now I wanted to switch to wss to have encryption, but...

This is a note to announce the availability of "Ogg Theorarm", an optimised ARM implementation of decoding libraries for the Theora video code, and Vorbis audio codec from xiph.org. Full details of this code release can be found at <http://www.wss.co.uk/pinknoise/theorarm>, but highlights include: * Full speed playback of a 320x240x25fps clip with a 48kHz stereo audio track on an 416Mhz Xscale based wince device. (39fps if audio sync is sacrificed). * With post processing (deblocking/deringing) enabled, this drops to 23fps. * Uses n...

Hi, I use chrome and sipml5 to connect to asterisk webrtc interface using TLS. The wss connection seems ok and the SIP REGISTER sent from chrome to asterisk and the SIP response received. In the response, I get a "failed: A server must not mask any frames that it sends to the client" error msg and chrome terminates the ws connection. I've checked the asterisk debug logs...

Hi I want to du a sensitivity analysis using Latin Hypercubes. But my parameters have to fulfill two conditions: 1) ranging from 0 to 1 2) have to sum up to 1 So far I am using the lhs package and am doing the following: library(lhs) ws <- improvedLHS(1000, 7) wsSums <- rowSums(ws) wss <- ws / wsSums but I think I can't do that, as after the normalization > min(wss) [1] 0.0001113015 > max(wss) [1] 0.5095729 Therefore my question: how can I create a Latin Hypercube whicgh fulfills the conditions 1) and 2)? Thanks a lot Rainer -- Rainer...

...s://123.123.123.123:8088/ws in Expert > Mode form (see [1]), I'm getting this error : > *2:SecurityError: Failed to construct 'WebSocket': An insecure WebSocket > connection may not be initiated from a page loaded over HTTPS.* > If I replace ws://123.123.123.123:8088/ws with wss:// > 123.123.123.123:8088/ws, this error message becomes with > *Disconnected: Failed to connet to the server* > > My questions are: > 1. Is wss now required by sipml5 live demo (implying wiki page is not > up-to-date) ? > > Yes, like the error says, you have to use wss on p...

Hi, I have a working WebRTC/SipJS+Asterisk(13.0.1) setup using ws sockets. Now I wanted to switch to wss to have encryption, but cannot find the required configuration parameters. Does Asterisk support wss sockets? How can I configure it? Thanks, Alexej -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/201...

Hi, I'm trying to connect to the asterisk pbx via wss, from sipml5.org demo page (http://sipml5.org/call.htm). I used the guide from https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial , to setup the tls. I could make a secure sip call ( SRTP) using the PhonerLite sip client. ( This confirms my sip - tls settings and tls certficates....

...ype something like ws://123.123.123.123:8088/ws in Expert Mode form (see [1]), I'm getting this error : *2:SecurityError: Failed to construct 'WebSocket': An insecure WebSocket connection may not be initiated from a page loaded over HTTPS.* If I replace ws://123.123.123.123:8088/ws with wss://123.123.123.123:8088/ws, this error message becomes with *Disconnected: Failed to connet to the server* My questions are: 1. Is wss now required by sipml5 live demo (implying wiki page is not up-to-date) ? 2. Do you have any pointer for WebRTC with Asterisk 13 and PJSIP ? Regards [1] https://w...

2016-02-18 14:57 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>: > > Is it implied here that both HTTPS and WSS must also come from the same >> server (Same Origin Policy) ? >> > No, the same origin policy does not apply to web sockets. > > Then, can I also install my own WebRTC demo page on my own private >> Asterisk server and access this demo page through HTTPS ? >> If I...

...fighting with webrtc for 14 days reporting my experience to minimize number of crazy asterisk users i have working webrtc with simpl5 + asterisk 13 + pjproject 2.4.5 + chan_pjsip + secure websockets + secure audio + audio in both ways problems first, i needed run chan_sip for old hard phones and wss with chan_pjsip only for webrtc. this is possible with patch from https://issues.asterisk.org/jira/browse/ASTERISK-24106 chan_sip is not usable for webrtc because of https://issues.asterisk.org/jira/browse/ASTERISK-24602 another problem arise with RTP/SAVPF negotiation this can be solved with pa...

Is it possible to use serveral protocols for a single transport section in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you cound use webrtc along with your phones but if I try: [transport-udp] type=transport protocol=udp,ws,wss bind=0.0.0.0 I get an error that transport-udp is not found. Do I need a dedicated interface for WebRTC? [Feb 15 12:42:10] ERROR[3308]: res_pjsip.c:2432 sip_get_tpselector_fro...

...erver or did you use Live demo ( https://www.doubango.org/sipml5/call.htm?svn=241) ? > Dne 18.2.2016 v 15:36 Olivier napsal(a): > > > > 2016-02-18 14:57 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>: > >> >> Is it implied here that both HTTPS and WSS must also come from the same >>> server (Same Origin Policy) ? >>> >> No, the same origin policy does not apply to web sockets. >> >> Then, can I also install my own WebRTC demo page on my own private >>> Asterisk server and access this demo page through...

....0.0 callerid=Usuario 1 elx4 <1000> callcounter=yes faxdetect=no [1001] deny=0.0.0.0/0.0.0.0 secret=ce93963b0751ed9a88ec1badbc073fce dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic trustrpid=yes sendrpid=no type=friend nat=yes port=5060 qualify=yes qualifyfreq=60 transport=wss,ws,udp,tcp,tls avpf=yes icesupport=yes dtlsenable=yes dtlsverify=no dtlssetup=actpass dtlscertfile=/var/lib/asterisk/keys/localhost.crt dtlsprivatekey=/var/lib/asterisk/keys/localhost.key encryption=yes callgroup= pickupgroup= dial=SIP/1001 mailbox=1001 at device permit=0.0.0.0/0.0.0.0 callerid=Usu...

...er(m), + as.integer(ncol(x)), centers = as.double(centers), + as.integer(k), c1 = integer(m), integer(m), nc = integer(k), + double(k), double(k), integer(k), double(m), + integer(k), integer(k), as.integer(iter.max), + wss = double(k), ifault = 0L) + switch(Z$ifault, stop("empty cluster: try a better set of initial centers", + call. = FALSE), warning(gettextf("did not converge in %d iterations", + iter.max), call. = FALSE, domain = NA), stop("number...

...dataS <- scale(d, center = TRUE, scale=TRUE) #Converts mydataS from a matrix to a data frame mydataS2 <- as.data.frame(mydataS) #removes "coden" variable myvars <- names(mydataS2) %in% c("coden") mydataSNc <- mydataS2[!myvars] #Determine number of clusters wss <- (nrow(mydataSNc)-1)*sum(apply(mydataSNc,2,var)) for (i in 2:15) wss[i] <- sum(kmeans(mydataSNc, centers=i)$withinss) plot(1:15, wss, type="b", xlab="Number of Clusters",ylab="Within groups sum of squares") #perform Kmeans with 2 clusters kmSNc2 <-...

...t (I can see myself), but never video on the receiving side. This is the case in both directions (so it makes no difference which peer is calling which peer). Both webRTC SIP peers have opus and H264 codec in their peer definition :   Video Support: Yes   Prim.Transp. : WS   Allowed.Trsp : WSS   SIP Options  : (none)   Codecs       : (opus|h264)   Status       : OK (75 ms)   Useragent    : SIP.js/0.12.0   Reg. Contact : sip:llghjqha at 192.0.2.239;transport=wss   RTP Engine   : asterisk   Encryption   : Yes   RTCP Mux     : Yes   Video Support: Yes   Prim.Transp. : WS   Allo...

On 2/15/16 12:50 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> Is it possible to use serveral protocols for a single transport section >> in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you >> cound use webrtc along with your phones but if I try: >> >> [transport-udp] >> type=transport >> protocol=udp,ws,wss >> bind=0.0.0.0 > > No, each transport is for a specific protocol. You can have multiple. > And how is this accomplished? If I...