search for: 500ms

...d invite obviously is interpreted as second, new call). Asterisk therefore cancels the first(!!) call - but Teconisy proceeds with exactly this first call (which now can't be handled by asterisk any more). For me, there are two problems in asterisk at this point: 1. The VoIP standard defines 500ms for t1 - using this standard value for t1min works fine with Teconisy, too. t1min should be always 500ms - it prevents a completely blocked line! 2. Why does asterisk stop the call completely after the second invite, which is canceled by Teconisy? It should be ignored because it means,...

http://bugs.digium.com/bug_view_page.php?bug_id=0001589 Has anyone else heard an audible blip, break or garble between answer and the native bridge attempt using sip? If I change the usleep(500000); to usleep(5000); in rtp.c the proble totally goes away... even the note above it says it needs to be fixed. bkw -------------- next part -------------- An HTML attachment was scrubbed... URL:

What is the best way to pick a prebuffering length for a streaming audio application using UDP transport? I'm using Speex in a VoIP application with RTP transport, currently with a fixed 500ms prebuffer on the playback side. However, I'd like something a bit more adaptive to accomodate high-jitter connections. For example, in one test configuration there is a very low average round-trip latency (50ms), but it spikes all over the place (sometimes 10ms, sometimes 500ms). Thus I c...

...However, every three seconds the CPU spikes to 50%. This is "system" utilization, not userland. I assume it's in a wcfxs interrupt. The number of interrupts stays constant at about 2004 during each spike, leading me to the conclusion that the TDM card is holding an interrupt for 500ms every three seconds (50% of 1000ms is 500ms). This is a disaster for spandsp and VoIP in general. When I unload the wcfxs module, CPU idle goes back to a constant 100%. The TDM22B card is REV E/F, and I've tried it with several different cards. Fedora Core 3 with linux-2.6.9 downloaded fr...

...ging. I'm not an expert but it seems like my asterisk doesn't recognize the hangup signal from the HT488 -or it's the HT88 which doesn't hangup upon the signal. this is my HT488 FXO config: PSTN Disconnect Tone: Frequency: f1 420 f2 420 PSTN Disconnect Tone Cadence: Choice 1: On 500ms Off 500ms Asterisk SVN-trunk-r48967 could someone please help? Thanks, Itamar.

...scussion Subject: Re: [Asterisk-Users] OT: Snom 320, displaying text on the screen from * I have that set, but for some reason I get errors when I try sipsak, and nothing comes through to the phone: sipsak -M -B "test" -s sip:44@192.168.1.67 <sip:44@192.168.1.67> timeout after 500ms timeout after 500ms... Some debugging info: [root@firewall root]# sipsak -vvv -M -B "test" -s sip:44@192.168.1.67 <sip:44@192.168.1.67> warning: ignoring -i option when in usrloc mode fqdnhostname: 192.168.1.1 our Via-Line: Via: SIP/2.0/UDP 192.168.1.1:34213;branch=z9hG4bK.105f...

...e mardi 14 juin 2005 ? 17:50 -0700, David Barrett a ?crit : > >>What is the best way to pick a prebuffering length for a streaming audio >>application using UDP transport? >> >>I'm using Speex in a VoIP application with RTP transport, currently with >>a fixed 500ms prebuffer on the playback side. However, I'd like >>something a bit more adaptive to accomodate high-jitter connections. >> >>For example, in one test configuration there is a very low average >>round-trip latency (50ms), but it spikes all over the place (sometimes &...

...nd what benefit it offers in reliable, high-jitter environments. So far as I can tell, the only solution to jittery transport is an adequate prebuffer, and thus I'm looking for advice on how to determine what "adequate" means. Likewise, I can easily broadcast anywhere from 33ms to 500ms audio packets (I currently use 50ms), but I'd like to hear your real-world advice on what the ideal packet size is I should be using. Thanks for all your help! -david Jean-Marc Valin wrote: > I strongly suggest you start by reading the Speex manual (you can skip > the technical parts...

Hi, What kind of latency is expected using 8,16,32 khz? I am trying to do a realtime stream server, and I am having latency above >500ms ( I capture the sound using the mic, encode it and send it to the client). I am using ALSA system, a SB128 PCI and a 800Mhz P3 . What can I do to lower the latency? I tried a test : arecord -t raw | speexenc - - | speexdec - and I found that I also have a 500 ms latency. <p>--- >8 ---- L...

...this case: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in UK and he is calling China, (japan have latency around 500ms to UK and 100ms to China), how asterisk will deal with this call?? Will his latency be JAPN->UK + UK->China (around 1000ms !) or only from Japan to China??? Please let me know. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digiu...

...hfsc rt dmax 2ms rate 256kbit ls dmax 3ms rate 690kbit #http int ${TC} class add dev ${LAN_IFACE} parent 1:2 classid 1:25 hfsc rt dmax 100ms rate 10mbit ls dmax 100ms rate 80mbit #squid ${TC} class add dev ${LAN_IFACE} parent 1:2 classid 1:30 hfsc rt dmax 200ms rate 80kbit ls dmax 500ms rate 80kbit #wan ${TC} class add dev ${LAN_IFACE} parent 1:3 classid 1:40 hfsc rt dmax 20ms rate 32kbit ls dmax 500ms rate 690kbit #msn ${TC} class add dev ${LAN_IFACE} parent 1:3 classid 1:50 hfsc rt dmax 200ms rate 32kbit ls dmax 500ms rate 690kbit # http bulk ${TC} cla...

Hello-- Why do I target chan_sip for so much effort? Because, it seems to me, chan_sip is probably the most used channel driver in the asterisk community!! (and, of course, the zap/dahdi driver, is also pretty popular) I haven't had time to follow up on chan_sip, and I probably won't for several months. But, if I had time, here is what I'd do: There are two ways to speed up

...ter.h Jean-Marc Le mardi 14 juin 2005 ? 17:50 -0700, David Barrett a ?crit : > What is the best way to pick a prebuffering length for a streaming audio > application using UDP transport? > > I'm using Speex in a VoIP application with RTP transport, currently with > a fixed 500ms prebuffer on the playback side. However, I'd like > something a bit more adaptive to accomodate high-jitter connections. > > For example, in one test configuration there is a very low average > round-trip latency (50ms), but it spikes all over the place (sometimes > 10ms, so...

Our new * server with Cisco SIP phones is now working pretty well. But we still have one problem that has not been resolved. The Cisco SIP phones have a consistent echo. It sounds like about a 500ms delay. This echo is not heard from the POTS side. It is heard when calling another SIP extension in our office. We are running 6.3 firmware on the phones. Any suggestions would be appreciated. Many thanks, Tom

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 organi...

...is equal to 1499. Moreover, any other value I try to configure in "b" unit is always decremented by 1 (3000->2999, and so on). Any clue about that? 2. I also have some strange behaviours about latency configuration - it is always far from what I request. If, for example, I try to set 500ms, the most probably I get something around 620ms. Note that if I configure the "limit" parameter instead of "latency", I can manage to reach values closer to what I need (e.g. 500ms). Is there something I''m missing about latency configuration? 3. I''ve tried to...

...one. I've tried turning on echocancel and echotraining in the Zapata configuration, but it had no effect. The website below states that two IP phones going through Asterisk should not result in a noticeable echo, instead this is more common when connecting to PSTN lines. The echo is about 500ms, faint but distracting. Any suggestions would be greatly appreciated. http://www.voip-info.org/wiki-Asterisk+Echo+Avoidance Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050526/2efefd33/att...

I am experimenting with an asterisk setup in my office. The last bit I have to test is working with analog lines. I have TE411p digium card, with an ISDN line plugged into the first, a channel bank plugged into the second port, and the last two ports empty. I have the following setup in my zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,d4,ami fxsks=25 And in zapata.conf, I

...give early randomness at probe() > time, and hence lose out on the opportunity to contribute to system > randomness at boot- or device hotplug- time. > > This commit schedules a delayed work item for such devices, and fetches > early randomness after a delay. Currently the delay is 500ms, which is > enough for the lone device that needs such treatment: virtio-rng. > > CC: Kees Cook <keescook at chromium.org> > CC: Jason Cooper <jason at lakedaemon.net> > CC: Herbert Xu <herbert at gondor.apana.org.au> > Signed-off-by: Amit Shah <amit.shah at...

...give early randomness at probe() > time, and hence lose out on the opportunity to contribute to system > randomness at boot- or device hotplug- time. > > This commit schedules a delayed work item for such devices, and fetches > early randomness after a delay. Currently the delay is 500ms, which is > enough for the lone device that needs such treatment: virtio-rng. > > CC: Kees Cook <keescook at chromium.org> > CC: Jason Cooper <jason at lakedaemon.net> > CC: Herbert Xu <herbert at gondor.apana.org.au> > Signed-off-by: Amit Shah <amit.shah at...