similar to: Question on Opus UDP payload

FYI, the Opus RTP payload format is now RFC7587: https://tools.ietf.org/html/rfc7587 Cheers, Jean-Marc -------- Forwarded Message -------- Subject: [payload] RFC 7587 on RTP Payload Format for the Opus Speech and Audio Codec Date: Tue, 30 Jun 2015 16:33:17 -0700 (PDT) From: rfc-editor at rfc-editor.org To: ietf-announce at ietf.org, rfc-dist at rfc-editor.org CC: drafts-update-ref at

Hello All When sending the Opus Packet in RTP Payload, the compressed frame is the output of the encoder? Also the config value as given in the RFC6716, 16...19 | CELT-only | NB | 2.5, 5, 10, 20 ms 16 corresponds to 2.5 ms 17 corresponds to 5 ms 18 corresponds to 10 ms 19 corresponds to 20 ms Is this correct representation of the data? Also in the RFC3551 the payload

Hi all, I am trying to compare the performance of gcc , llvm-gcc , clang and lli(with JIT) on x86. i have attached the performance comparision spreadsheet as well as the source which i used for performing these test. i ran this code for 10000 iterations and the time of execution is as follows for -O3 results refer attachment. *clang (-O0) * real 0m10.247s user 0m2.644s sys 0m5.949s

Hi All, I am doing some testing with the asterisk server and have been monitoring the packets exchanged during a SIP-ZAPTEL phone call. I see that the IP-ID in all of the RTP/UDP/IP packets are set to zero. After some googling, I have learnt that some of the linux implementations set the IP-ID to 0 (if the DF bit is set in the IP header) if the two hosts exchanging data are on the same subnet.

I'm currently implementing A rtp header info and using speex for audio. can someone tell me what the rtp payload type value is for speex. i've checked the submitted speex draft and it says its beyond its scope and i can't find anything in RFC3550 that might suggest what it is. any help would be appreciated thank you, Greg --------------------------------- Boardwalk for $500?

hmm...forgive my ignorance here. icould have explained it wrong. the rtp header has the pt (payload) field as a 7 bit value. i was under the impression speex had a particular value i should set it to. is this so? if no what value should i assign it, whether by convention or otherwise? Note that i'm implementing a simple rtp header and combining it with the speex payload i'm not using

Question for Everyone, We are running 1.9.18p8 on hpux 10.20. This particular machine has a directory called docommon that is writable by anyone. The problem we are having is that whenever someone opens a file with word 97, then saves it, they own the file and the permissions are -rw-r--r--. Thus, no one else can open the file. How can we make the files keep the -rw-rw-rw setting? Please

re-posting the question. ----------- use case: when some one in my pbx calls 100.200, I have translations well defined, Media also (media via asterisk) --Works. when some one calls bob, or for any names I am adding Domain and call is been sent to the other party -- Works, no media... For the cases when it is talking to the external work, I want Astersik not to do anything with the SDP/payload.

(hello all, this is my first writing. so please bear with me if I'm wrong anywhere.) orry to break too lately, but how is the RTP payload submission is going? could we see the new payload at March IETF? I agree that it would be fairy straightforward to make an RTP payload for ogg vorbis, assuming raw packets, AFAIK. using physical bitstream is, in this case, not adequate by the reasons in

I am about to connect to a new provider who requires 20ms payload sizes in g729a. Is this configurable on asterisk? Is 20ms the default? Cheers, j

Hello, I encountered the dtmf problem between my asterisk box (1.4.23) and suppliers gateway (unknown vendor). I have dtmf mode set to rfc2833 and it alway worked till supplier has changed something. Now I receive from him dtmf payload 100. With the second supplier which sends dtmf with payload type 101 everything works. in cli I get this message as dtmf is entered rtp.c:1287 ast_rtp_read:

Hi, I was womdering, if anyone has experienced the same issues. When I run the tests after compiling 2.3.5, the following 4 tests fail: http payload echo (ssl): sequential .................................. : ok http payload echo (ssl): pipeline .................................... : ok http payload echo (ssl): parallel .................................... : FAILED: Test is hanging http payload

Hello. It appears that I have an issue with my dovecot setup. When trying to authenticate, i get this error: May 28 20:18:26 AmaechiJ dovecot[10516]: imap-login: Login: user=< aj at example.com>, method=PLAIN, rip=, lip=, mpid=10879, TLS, ses...QOQBg9rlR> May 28 20:18:42 AmaechiJ dovecot[10516]: imap(aj at example.com): Connection closed in=116 out=1311 May 28 20:20:32 AmaechiJ

Hi All, I have a SIP provider that tells me that my RTP stream uses a "20bytes payload in the g729 coded data". And they would like that we change this to 30bytes (3 frames). But maybe I'm wrong but isn't a certain payload just a standard for a codec ? And if I'm wrong, how can I change the payload for my g729 calls in Asterisk. Greetings, Attilla

I can't seem to get musiconhold to work. I'm running asterisk on a RH9 box, I have the mpg123 package installed. In my zapata.conf file I have the line MusicOnHold=default . In my musiconhold.conf file, in the classes section I uncommented default and loud. In my extensions.conf file I have a set musiconhold line. However if I get a call and I either put it on hold or hit flash

Hello to all speex developers, I have question regarding payload length of narrowband speex in RTP. I were watching tcpdump of the xlite softphone and have found that it uses weird payload length namely 75 Bytes I went through various source and without success. To be clear: For 8000Hz sample in 20 ms that is 160 samples per frame. This makes 50 frames per sec. modes bit-rate 8 kbit/s

Hi >i have aproblem withe execution of my function >first, i wrote my function in the script of R >nom_fonction <- function(arg1[=expr1], arg2[=expr2], ...){ bloc d'instructions } > when i want to have the result i mean the laste instruction in the bloc of > instruction , i try to >wrote the name of function >source(aj.fun) Error in readLines(file, warn =

Hi, Has anybody some information on on the "Redundant audio data" header in the speex rtp payload? Is this header always present? Is its value always the same? Can it be modified through some speex_*_ctl function? Thanks, Emmanuel -- ------------------------------------------------------------- Emmanuel Wauters Tel : (+32) 11 30 13 30

HI >i have a problem please help me to solve it: http://r.789695.n4.nabble.com/file/n4639434/aj.pdf aj.pdf >i want to calculate the vecteur a[j] where j: 1...8 >this is the code in R: >aj.fun <- function(j, i, X, z, E, beta0, beta1){ + n <- length(X) + iX <- order(X) + iz <- order(z) + e1 <- -(beta)*z[ iz[1:(i - 1)] ] + numer <- E[j] - sum( X[ iX[1:(i - 1)] ]

While trying to use direct_media I'm seeing RTP payload mismatches after succesful reinvites. Initial INVITE from endpoint A to asterisk has rfc4733 DMTF m=audio 35648 RTP/AVP 9 8 111 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 >From asterisk to upstream U: m=audio 14338 RTP/AVP 9 8 111 18 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 So the payload types in the RTP