similar to: About speex quality

blink : It use iLBC also. Voice over IP RTP: A Transport Protocol for Real-Time Applications RFC3550 RCTP: Real Time Control Protocol attribute in Session Description Protocol RFC3605 SRTP: The Secure Real-time Transport Protocol RFC3711 DTMF: Dual-tone Multi-frequency Signaling RFC2833 and inband MWI: Message Summary Event Package RFC3842 Speex and G722: Wide-band Internet Codecs G711, iLBC

We have no feedback or information about iLBC usage, is almost not used at all looking at our usage charts. Adrian On Apr 12, 2010, at 3:44 PM, messaging bay wrote: > blink : It use iLBC also. > > Voice over IP > > RTP: A Transport Protocol for Real-Time Applications RFC3550 > RCTP: Real Time Control Protocol attribute in Session Description > Protocol RFC3605 >

I. most of ssh manual and all sshd manual present server and client as one machine, called host. All files mentioned are placed on one machine. This is incorrect, and makes the explanation unclear. For example, man sshd SSH_KNOWN_HOSTS FILE FORMAT suggests to copy keys from /etc/ssh/ssh_host_key.pub into /etc/ssh/ssh_known_hosts, as if those files are on the same machine. II. a general

The short: Linking to libspeex.a when generating a .so using libtool results in a non-portability warning. This is due to PIC code and non-PIC code intermingling. How can I go about fixing this whilst still using an installed libspeex present on the user's system? The long: I am using autoconf + libtool to generate a codec plugin for speex (sipXmediaLib), and I'm trying to eliminate

Hello;I just came across NUT, testing FreeNAS/FreeBSD. I?m running at home a APC UPS connected to another computer, running apcupsd. I found out that NUT didn't talk by default to apcupsd, and there was a driver created relatively recently (apcupsd-ups), which filled in that gap. The version I found on my FreeBSD/FreeNAS system seems to imply that there is more to come to that driver (0.04)

Hi, are there any open source DRM implementations for protecting Theora streams over RTP? I only know of the ISMACryp specification which seems to be licence free and so could be used in an open source project. Regards, Franz

Hello, I heard that myJoice used Speex as the audio codec. Has anyone experienced its performance? By the way, is it possible to buy it outside of Sweden? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20100508/1a1eefb1/attachment.htm

>> Yes, I plan to use it in a VoIP environment if I can get latency reduced to >> an acceptable level :) >> The latency depends directly on the overlap parameter, which also controls >> the quality. Higher quality => higher latency. You could set the overlap to >> 0, but that would give you some nasty artifacts. >> You can also resample with smaller block

Inspired by AG Projects' Adrian Georgescu's post of Eric S. Raymond's classic "How to Ask Questions the Smart Way" to the OpenSIPS-users mailing list[1], I'm going to repost it here: http://www.catb.org/~esr/faqs/smart-questions.html As Adrian said, "This a good read for those who show up on mailing lists without any guidance about how to ask the right

FYI, for those of you who aren't on the serusers list. I'd like to hear how others can get this working in small Asterisk settings; I don't really have the time to implement it, but it looks very interesting. JT >To: serusers@iptel.org >From: Adrian Georgescu <ag@ag-projects.com> >Date: Mon, 31 May 2004 23:05:47 +0200 >Subject: [Serusers] CDR mediation for VoIP

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

Hi, I'm excited to announce that the guys from AG Projects are stopping by for a beer tomorrow on VoIP Users Conference, aka VUC. You should already be familiar with their excellent multi-platform SIP client, Blink (http://icanblink.com) While Adrian and Sa?l enjoy a few exotic brews with us, they'll also be telling us about SylkServer: - creation and delivery of rich multimedia

MessageMelissa - I added the "fromuser=AnyName" to my sip.conf file stations and that, in fact, corrected the problem. The MWI now works flawlessly. I would recommend that Aastra/Sayson pursue this with the Asterisk team so that it is listed as a known issue or to have Asterisk patched to fix it. I will submit a bug on it as well. I'm copying the Users Mailing List on this. For

Ok. I did some quality tests. First off; never do quality tests with ints. I had serious problems interpreting my results until it dawned on me that the signal differences were just 0 or 1. So, after a lot of scratching my head, these are done comparing the result from the _float versions (which is how both resamplers work internally anyway). What I did was this: Load speex_wb.wav as one