search for: 700ms

Hello everyone, Can any body please tell me the compiler options to optimise the speex code for ARM TE architecture. Fixed point is enabled. It currently takes about 700ms to encode 5 sec worth of voice. I need to get it down to under 10ms. Is the option --enable-arm5te-asm supported? I am using speex 1.1.12 version. The configure.ac file does not seem to support --enable-arm5te-asm option. I tried --enable-arm5e-asm but the result is'nt much different. Re...

Hello, I have 2 sites. One(Site 1) has an asterisk PBx and the Other(site 2) has 2 remote soft phones. The latency btw both sites is btw 500ms-700ms. I know this is a shot in the dark...but are there ways of improving the voice quality for the remote calls(btw site 1 and site 2), Other than increasing bandwidth? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization...

...-Marc Naveed Rehman a ?crit : > Hi JM, > > I am using narrowband mode with speex qaulity set to 0 and complexity > set to 4. The code works fine on the 3.2GHz processor. The test > program speexenc takes 7ms to encode 5sec worth of voice on the > 3.2Ghz processor. The same takes 700ms on the ARM5E intel Xscale. The > code that I wrote puts 10 speex frames in one RTP packet. When > exexcuted on ARM I get lots of underrruns hence noise. I was > wondering perhaps its because of the speed of ARM processor as > dictated by the significantly slower speexenc on ARM. I hav...

...g I'd solved my CND (caller id) problem. My callerid.c change got it working on one of the handsets I have, but not the other. (Recap: The AU CND standard has an increased delay between the end of the ring signal and the start of the CND FSK data -- it's 500ms in US GR-30-CORE, and around 700ms in AU DC.002/TS030). I've manipulated this delay value in an attempt to come up with something that it likes, to no avail. This handset is a DORO 5045[1], and CND works fine when it's plugged into a telco line with CND enabled. It also works fine when attached to an ATA-186 with the def...

Well, Once upon a time, I had problems receiving callerid, and then one day, Mark was logged into my asterisk box helping with something else, and I asked him about this, and he showed me a nice tweak to some source file that made it work. Some time later, I must have done hundreds of CVS updates, and along the way, lost the above patch (one liner) and so callerID hasn't worked for a long

...are saying while we can hear them crystal clearly. Various pings and other network tests indicate that the underlying network is functioning as well as can be expected for a sat link. In fact, the overall jitter seems to be pretty low (avg 20ms). Packet loss is around 1-2%, and latency is around 700ms on average. I'm left to assume that the jitter buffer on that end isn't functioning properly. Both ends of the call have the same jitter buffer settings. The call is carried by IAX2 and encoded with ILBC. The iax.conf files on each end start like this: > [general] > t...

Hi JM, I am using narrowband mode with speex qaulity set to 0 and complexity set to 4. The code works fine on the 3.2GHz processor. The test program speexenc takes 7ms to encode 5sec worth of voice on the 3.2Ghz processor. The same takes 700ms on the ARM5E intel Xscale. The code that I wrote puts 10 speex frames in one RTP packet. When exexcuted on ARM I get lots of underrruns hence noise. I was wondering perhaps its because of the speed of ARM processor as dictated by the significantly slower speexenc on ARM. I hav'nt gone much...

> Can any body please tell me the compiler options to optimise the > speex code for ARM TE architecture. Fixed point is enabled. It > currently takes about 700ms to encode 5 sec worth of voice. I need to > get it down to under 10ms. The ARM port has suffered a bit from the fact that I haven't been able to keep the assembly up-to-date. Don't know what bit-rate/options you're using so it's hard to say much about the absolute numbers. The o...

Is any one using a Direcway Satalite to stream audio out. I want to do some live broadcasting next month and I live so far out in the country that this is the only broadband source available. Is there any source of information on this to read up on? Any problems anyone has run into? Things I should look out for or do? Thanks in advances. Dave Kelly

Does anyone have any insignt on this? Any client programs that could be used? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030925/0e9e9920/attachment.htm

Hi, I have two analog phones connected on the digium tdm22b; I can't put calls on hold by pressing the R button on the phone. I can do it only by hook flash. How can I configure asterisk to use the R button? Thank you in advance.

We have a tenant who has been having issues with a congested connection and in trouble shooting it we've noticed that there seems to be a lot of SIP traffic even when none of the phones are doing anything. We've determined that this traffic is mostly INFO packets generated by setting qualify=2000. I understand that 2000 ms is the default value for the qualification parameter but what

Has anyone from digium looked at why there is a 30 min to 3 hour lag on messages on this list? I.e looking at the last 50 messages I've received, the lag is about 90 minutes between the time sent and the time received. Sometimes this drops to as little as 4 minutes. Is this problem worse for me because my email address starts with "w" and my copies of the emails get sent after

Hello, I have an Asterisk with one local Cisco ATA and one remote Cisco ATA connected to the Asterisk, the remore connection is a satellite link with an 900ms delay. I can make calls from the remote site to the local site, but when try to call from local to remote it doesn't work. The Asterisk timesout, it sais no one answered and can?t establish the connection. Can anybody help me with

Hi All; Can I run the IAX on TCP port instead of UDP port? If I ran IAX in TCP port, and in case my network was having a lot of users doing browse on the internet and downloading, so in that case and if the IAX used TCP port, so the voice will be better than using UDP (because in TCP the lost packets will be resend while in TCP it will not which will cause the voice to be cutting)? Same thing

> On Sep 14, 2015, at 5:02 PM, Mehdi Amini via llvm-dev <llvm-dev at lists.llvm.org> wrote: > >> >> On Sep 14, 2015, at 2:58 PM, Pete Cooper via llvm-dev <llvm-dev at lists.llvm.org> wrote: >> >> >>> On Sep 14, 2015, at 2:49 PM, Matt Arsenault via llvm-dev <llvm-dev at lists.llvm.org> wrote: >>> >>> On 09/14/2015 02:47