similar to: Doubts about the Monitoring command

Hi all, I would like to make a call using the asterisk IAX with g726 - 16 bits codec. How could I configure it in the iax.conf file. Do I need to modify the file like this? . . disallow = all allow = g72616k . . I have tried it but it hasn´t worked. Thanks in advance and best regards Guild __________________________________ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today!

Hi, We are planning to use Speex as the speech codec for a VoIP application. 1) May I know how Speex compared with GIPS codec? It seems that Google, Yahoo, and Skype are licensing from GIPS. Are there any good benchmarking or fair comparisons? 2) In particular, how is the jitter buffer control for Speex in response to intermitent poor connection hiccups? Is it robust enough to smooth out

Daniel Schmidt a ?crit : > I found the cause of the problem. The function shift_timings can > produce overflows in the timing array if the jitter is huge or the > time units are very short. After changing the timing values' type from > spx_int16_t to spx_int32_t it seems to work. Hmm, I always assumed there wouldn't be any overflows. What parameter range are you using that

Hi, I'm a developer currently using the speex codec in a VOIP application of ours.. It sounds amazing, especially at wideband.. my question is how do I force it to do a concealment? We have a low latency application, and based on the current API, I'm guessing concealment only kicks in when a packet is lost.. However, our jitter buffer knows when a packet is missing, and I'd like to

Okay, this is really simple but because Google has deteriorated in quality over time, it's getting really hard to find straight forward answers. Very simple; when does the %POST section of the Kickstart file kick in during the install process? Before or after the first reboot? Hopefully this gets picked up easy by the spiders... -------------- next part -------------- An HTML attachment was

Hello I've been using VoIP over a not so reliable net: I usually get a 5% to 10% packet loss and a very high jitter. I tried several codecs and parameters, and the only thing left to test is PLC (Packet Loss Cancellement). Have the astesrisk and digium people implemented PLC?, Are they implmementing it now? and, if not, Where can i find an implementation? Thanks in advance -- Jorge

At one point I did have this kit working but since upgrading to the latest Asterisk, it no longer seems to. I had the following problems after several reinstallations: - USB adaptor had a proper dialtone, asterisk recognised the pickup, but pressing keys on the handset had no effect - USB adaptor produced a strange horrible tone, not a dialtone; pressing keys has no effect - USB

Hello, I'm experiencing some problems with running IAX2 protocol on quite reliable link with G729A codec. My customer has 2mb FR link to the Internet used in about 20%. Ping statistics: 50 packets transmitted, 49 received, 2% packet loss, time 49496ms rtt min/avg/max/mdev = 9.308/13.126/33.307/4.851 ms Everything would be great, but the quality isn't good enough. I have 2mb/512kb DSL

Hello, We have 15 Soundpoint IP 500 phones and recently call quality has deteriorated. On some calls there is a static-buzzing of sorts that occurs when users talk. It can be picked up on SIP-SIP calls and SIP-ZAPTEL (Channel BANK<<-T100P). It basically sounds really weird whenever someone talks, it sounds like a bee buzzing or something. Very hard to explain. Also, there will be echo

Hi All I have recently downloaded Asterisk and was so impressed I thought I would setup a home server and I went out and got myself a couple of cisco 7940's. (and a sipaura 3000!). thanks to various posts on this list and the voip-info site I have managed to get chan_sccp setup and working with the 7940's but the I tried to get the messages, services and softkeys working. It seems

Hi, I found this message concerning VAD and was wondering whether VAD has been ported to fixed-point in the latest version? Thanks, SingHui ---------- Forwarded message ---------- From: Jean-Marc Valin <Jean-Marc.Valin@usherbrooke.ca> Date: Jul 22, 2005 1:02 AM Subject: Re: [Speex-dev] Fixed-point To: gue baja <gue_baja@yahoo.com> Cc: speex-dev@xiph.org Hi Baja, Here's a quick

We're investigating audio quality issues in our system; maybe someone can help. We're using Asterisk as a basic PBX, with a single PRI on one side and SIP phones on the other: Sipura SPA-841's. We're experiencing several audio effects which seem to commonly correspond to network failures (packet loss, high jitter, etc manifested as "robot voice", dropouts, periodic

> (PS, if you do use threads, protect speex_jitter_put/get with a mutex > (CRITICAL_SECTION I believe they're called in Win32Speak) -- calling put > and get at the exact same time from different threads leads to "features") I've never tested this, but I designed the jitter buffer to work from two threads even without using a mutex. This would work as long as there is

I need to debug a call quality issue with remote users on the other end of a satellite link. The symptoms are: we here on the Internet side can hear them just fine. On their end, things work sorta OK most times, but they often suffer from severe dropouts and digital warbling, both of which I attribute to them missing packets. Often times they can't make out a word we are saying while we can

> Obviously this is Jean-Marc's decision and I'm not telling > him not to support this compiler. I am however pointing > out that this compiler is yet more work for very little > payoff. In the case of my project, the proponent of C++ > Builder sent me a huge, monsterously ugly and totally > unmaintainable patch to add C++ Builder support. Needless > to say, that

I like the NCoC approach, actually. I think the community can manage itself quite well. Echoing a prior response---if this community really needs a set of rules, then it has already deteriorated. We are not there and it doesn't look like we are going in that direction. The FreeBSD CoC also looks reasonable, and if any CoC is needed, I find this one more appealing. -Krzysztof On

Hi, Ok, what you suggest sound logical to me. Currently, I have done a small trick to prevent this problem. What I did is that whenever windows request a voice packet from me and if I do not have the voice packet, I repeat the previous packet. Hence, all the breaking portion is filled with previous packet. This trick seems to work so far. I am not sure what is the side effect. I think jitter

Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I don't really know where to start on measuring jitter

Hello All, MOS and R factor are the two QoS parameters used to estimate VoIP call quality. I have found that they are calculated from other metrics like jitter, latency, packet loss,...etc. But, haven't found any formula or arithmetic rule to calculate them. Do you have an idea about their formulas or an open source that calculates them. Is it possible to interpret them from wireshark.

Hi Who can explain its return values and packets losing relationship during decoding? Because the decoder needs to consume one 20ms frame , if there is packages lose , the jitter buffer will be empty. Is it good idea to replace as silence? Regards Bay