search for: packetloss

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

....22) 03:24:17 : H 3 : I 19 : P 50 : cisco_snmp_ping_wait(): -1 -> no_backend(): 0 (time P:0.19 | 0.08) 03:24:17 : H 3 : I 21 : P 50 : snmp_counter:load_average_1(.1.3..3.1): 0.21 -> buffer(): 32 (time P:46.3 | 0.22) 03:24:17 : H 3 : I 18 : P 55 : cisco_snmp_ping_get_pl:packetloss(): 0 -> buffer(): 33 (time P:0.43 | 0.17) 03:24:17 : H 3 : I 19 : P 55 : cisco_snmp_ping_get_pl:packetloss(): 0 -> buffer(): 34 (time P:0.19 | 0.15) 03:24:17 : H 3 : I 21 : P 55 : snmp_counter:tcp_passive(.1.3..6.0): 26279 -> buffer(): 35 (time P:46.14 | 0.28) 03:24:17 :...

Hello speex team, Where can I download test samples (*.wav) from? I could play the samples from http://www.speex.org/samples/, now I want to encode and decode few samples myself. :-) I want to encode and decode them with all possible options (VBR, VAD, DTX, COMP, NOENH, PACKETLOSS, etc). Speex is wonderful. Thank you. :-) -Basawaraj Send instant messages to your online friends http://uk.messenger.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20061128/854c4fcc/attachme...

...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 on Cisco 7960. Any ideas on fixes or what may be causing this problem? Thanks, Sam Bacsa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040414/58b4...

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 to search for that Problem, but couldn''t find a really matching...

...s I can't hear the person talking othertimes they can't hear me. This situation comes and goes throughout the call. Bandwidth isn't an issue as I have a 3MB/1MB connection and there is at most 2 concurrent connections. Also using pingplotter to monitor iax2.sixtel.net shows little or no packetloss. Just as further info, I am using a SPA-2000 to connect to * with G711u as the preferred codec. Anyone else experience the like or have any suggestions on what may be causing this or ideas on how to debug? Brian

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

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

...it's dead on but alot of times it will see a single keypress as multiple keypresses. So I may press 561 but it will see 566661 and all three keypresses are about the same length. Is this unique to my case or do you others see this too. I suspect it's due to either background noise or maybe packetloss? Any ideas on how to clean this up? 2. The only way I can get the app to fire off is if I put the extension mapping in as _NXXNXXXXXX,1,CMD I'd like to use s,1,CMD but I don't know what I'm missing here or doing wrong. Below are a copies of my extensions.conf file and my iax.conf file...

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

...packet loss(like RDP application, I guess if the packet loss go to certain threshold, the connection will be lost). TINC/OpenVPN/IPsec/L2TP/<insert VPN tech of choice> should *NOT* be the place where you handle your network connection reliability If the upper level app/etc. can’t handle the packetloss(es), then you’ll have to either fix your network, or the upper level application, as TCP/IP already does retransmissions with packet losses, and since it’s just encapsulated over the vpn-tech-of-choice, it’s not the VPN-tech-of-choice that should retransmit, but the TCP/IP stack. > >> O...

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 [mailto:speex-dev-bounces@xiph.org] F?r Maurizio Cembalo Skickat: den 13 april 2007 18:30 Till: speex-dev@xiph.org ?mne: [Speex-dev] Symbian and buffer of 4096 bytes I...

...- where rtp also goes through the pbx - that don't use the PRI's) I tcpdumped all traffic on various points in the network between the phones and the pbx and noticed that the gaps are already present on the ethernet interface of the pbx, so I ruled out network problems. (It's not packetloss, just silence - I converted the pcap files to audio files and listened to it) Next step is to see whether the gaps originate on the Asterisk pbx or the external machine, so I wanted to dump traffic directly from the zaptel interface. The supplier of our pbx suggested that it may be possible...

...o make it work all the time, not just sometimes. What does the error message mean? Does "the user" mean the user I am logged in on NEPTUN or the remote user I am putting into the form when connecting the drive? Does "this computer" mean NEPTUN or OBERON? The network is OK, no packetloss. Cl<

...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 Chua <kelchy at gmail.com> To: Dragos Oancea <droancea at yahoo.com> Cc: Benjamin Schwartz <benjamin.m.schwartz at gmail.com>; "opus at xiph.org" <opus at xiph.org> Sent: Wednesday, March 4, 2015 11:02 AM Subject:...

...le smb1 again in windows 10 again. I noticed ms > changed things again. > Thinking here that the "older samba" your using, with a > latest windows is the problem. > Enable smb1 again, think that will fix a lot. > > And your sure you vpn line is ok and you dont have packetloss? > Think in test with mtr or smokeping, something like that. > > Are the MTU sizes are handled by the firewall? > This is to prevent IP packet fragmentation, so IPTables is > set to reduce the size of packets by adjusting the packets' > maximum segment size. > Somethi...

...0.10) these are the results from that test: --- 172.30.10.10 ping statistics --- 40707632 packets transmitted, 40694228 packets received, +4362 duplicates, 0% packet loss round-trip min/avg/max/stddev = 2.241/2.967/310.331/2.502 ms Now, as you can see, there really doesn't appear to be enough packetloss to warrant those errors every single time I connect to samba from the WinXP machine. There's also a Win98se machine on my LAN that does not get those errors. I can still use the shares as if there were nothing wrong. However, I would like to get to the bottom of this and see if it can'...

...e last 3 things. First try enable smb1 again in windows 10 again. I noticed ms changed things again. Thinking here that the "older samba" your using, with a latest windows is the problem. Enable smb1 again, think that will fix a lot. And your sure you vpn line is ok and you dont have packetloss? Think in test with mtr or smokeping, something like that. Are the MTU sizes are handled by the firewall? This is to prevent IP packet fragmentation, so IPTables is set to reduce the size of packets by adjusting the packets' maximum segment size. Something like this: iptables -A PREROUTING...

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