similar to: speex samples required

Hi all, It appears that something very strange is going on... Here's the deal -- whether someone calls in to my * server or I call out (doesn't matter), I can hear them perfectly: no gaps, no packet loss, nothing ... however, when I speak there seems to be very noticable latency and "choppiness" as if there were packetloss or lots of jitter. I'm using SIP for outgoing

I am running * 1.0.5 and have been having lots of problems with outgoing calls and their sound quality. I am using ULAW for the codec and sixtel for termination. Basically the problem is that portions of the call seem to be lost and replaced with silence. Sometimes I can't hear the person talking othertimes they can't hear me. This situation comes and goes throughout the call. Bandwidth

Hi there, I support a large number of enterprise users who contractually must connect to our support center via a 4G VOIP connection. I simply want to be able to auto detect all poor quality calls in realtme (as they are being made), play a message and drop the call - without user intervention. All decent call quality calls will be allowed through - to be handled by support staff. Its a

Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will read it back to them, if they press pound it will attempt to connect via the second IAX provider,

Hi all! I know this is quite offtopic, but I found nothing in google nor in the net-snmp mailing list, and I know there''s a lot of people here graphing with snmp+rrdtool. I installed net-snmp 5.1.2-6.1 in Debian to produce graphs for some ethernet interfaces. It seems it''s not reporting correct values for them. Here''s my snmpd.conf (very simple) and the output from the

Hi, I''m new to XEN and have the following Problem: If I shutdown my domU the network of the dom0 leaks. A ping from another server to my xen dom0 shows a Packetloss > 80%. Then if I restart the domU after some seconds the ping show a loss from 0%. I retryed that 3 times but it is really hard to go to the dom0 via ssh and recreate the domU if the packetloss is that high. I used google

Hi there, I have two ethernet connections of 2Mbit/s each and I''m trying to add them together to one 4Mbit/s connection but I cannot get more than approximate 1Mbit/s! My setup: I have a LAN (10.2.18.0/24), connected to a larger network (10.0.0.0/8) by two WAN-connections with 2Mbit/s each. On each end I have a Linux router. I first setup the routers to use TEQL with one of the

Hi I am looking for quite a while now to run Puppet inside a vserver. The installation worked perfectly, but when I run a test I get this message: smb-configmaster / # puppetd --server 192.168.0.144 --waitforcert 60 --test /usr/lib/ruby/site_ruby/1.8/puppet/network/client/ca.rb:31:in `request_cert'': Certificate retrieval failed: Could not connect to 192.168.0.144 on port 8140

I'm using speex under symbian (8000 hz, 16 bit) narrow band. The phones API only give me a buffer of 4096 bytes in recording.To reproduce audio I must fill up the buffer of the same dimension. 4096 isn't a multiple of 320. I want encode the audio in streaming. The solution that I adopt to encode is: - Divide 4096-256 bytes in 12 frames of 320 bytes. - Therefore the frame number 13 is

I agree with the in-effective of TCP transmission, but I wonder if the the UDP packet is dropped, the tinc VPN itself wouldn’t retransmit, and if the upper level application doesn’t handle the packet loss well, will this be the problem? Or the upper level application have very limited tolerance to packet loss(like RDP application, I guess if the packet loss go to certain threshold, the connection

Thanks Dragos, I assume I will be setting those parameters during initialization of encoder right? Question is, if connection gets too lossy, how will opus adapt to it? Can it automatically shift bitrate down to minimize impact? Mark from IRC suggests that the app has to be aware of the losses and change it on the fly. Has anybody on the list tried this? Kelvin Chua On Wed, Mar 4, 2015 at 5:53

Hello all, Since a while some of our SIP users complain about gaps (sometimes multiple seconds of silence) in the RX audio stream (direction pbx -> phones). Our configuration is an Asterisk with two Wildcard TE410P cards that are connected with E1 PRI's to an external server running callcenter software. The SIP users use Snom and Polycom phones. The gaps are not present in

> On 18 Jun 2017, at 15:44 , Bright Zhao <startryst at gmail.com> wrote: > > I agree with the in-effective of TCP transmission, but I wonder if the the UDP packet is dropped, the tinc VPN itself wouldn’t retransmit, and if the upper level application doesn’t handle the packet loss well, will this be the problem? > > Or the upper level application have very limited tolerance

Hello I have a Samba PDC in domain KEVF_D4 called OBERON and a NT4 workstation NEPTUN in workgroup (not domain, workgroup) WORKGROUP I tried to map \\oberon\linux from OBERON using smbclient \\\\oberon\\linux -U <username> and gave password and it worked. I tried to map \\oberon\linux from NEPTUN and 1) got error message: "\\Oberon is not accessible. Logon failure: the user has not

I would consider another solution to your problem. 1) Save the 256 bytes, wait for 64 more in the next 4096 buffer and give speex a complete frame. Or... 2) Not sure on this but I believe simply dropping the 13:th frame could be an option. < 10% packetloss is barely audible with correct decoding and playback. //JT -----Ursprungligt meddelande----- Fr?n: speex-dev-bounces@xiph.org

Hi Kelvin, The audio bandpass setting is only configurable when the encoder is instantiated (eg: start of a Voip call ) , but you can change the bitrate anytime.For example if you can read incoming RTCP packets , you can check if there's reported packet loss , and then lower the bitrate accordingly.Yes, the app has to be aware of the packetloss ?percentage. Cheers,Dragos From: Kelvin

Ps. Smb1 in Win10 ! I mean ofcourse.. Step 1: Go to Start button and type Control Panel. You will get Control Panel; Desktop app. Click on it. Step 2: You will get a window with a list of software on it, go to the left side of the panel and click Turn Windows Features On or Off. Step 3: You need to remove the checkmark beside SMB 1.0/CFs File Sharing Support to disable SMB1 for good. If you

I appologize if this has been answered previously or if the answer is obvious. I have done the best I could searching through the mailing list archives and through google to see if I can find an answer to my problem. With that said, the problem is that whenever I use my connection to samba (3.0.23c on FBSD 5.4) from my Windows XP (Professional, SP2) machine I receive these errors: Sep 11

Hai, > -----Oorspronkelijk bericht----- > Van: samba [mailto:samba-bounces at lists.samba.org] Namens > Stephen Davies via samba > Verzonden: donderdag 25 april 2019 8:34 > Aan: samba at lists.samba.org > Onderwerp: Re: [Samba] User mapping/login issue > > > > It would appear that there may be more than one issue with my smb.conf. > The scenario is a Centos 7

Hi - I am trying to find a Linux based solution for one of my clients, I want to bond two adsl lines into one, with redundancy (if one fails the other does all the work). I''ve been looking into ECMP (Equal Cost Multipath), I know BGP would work, but something like ECMP would be much simpler. I have full control of both ends of the connections and I can have both ADSL lines terminate in