similar to: GSM to iLBC one way audio :-(

Hi All, I've been playing with GSM and iLBC over low bandwidth connections (central Asterisk box with 2mbps, to ADSL users on 512/256) and both seem to perform well. Based upon what I've read in the archives and at voip-info.org iLBC should perform a little better if packets are lost, than compared to GSM. Do you find this to be true in practice, or is GSM just as robust? Whilst

Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no

Hi All, I have just installed Asterisk and am trying to connect it to a SIP account that I currently have with www.voiptalk.org but without any success. Although I know that voiptalk do provide asterisk accounts I don't want to convert the SIP account until am happy that it's gonna work for me. The asterisk box is currently behind a firewall and the following ports are being forwarded

Recently, we were notified that the mechanism included in our Asterisk source code releases to download and build support for the iLBC codec had stopped working correctly; a little investigation revealed that this occurred because of some changes on the ilbcfreeware.org website. These changes occurred as result of Google's acquisition of GIPS, who produced (and provided licenses for) the iLBC

Recently, we were notified that the mechanism included in our Asterisk source code releases to download and build support for the iLBC codec had stopped working correctly; a little investigation revealed that this occurred because of some changes on the ilbcfreeware.org website. These changes occurred as result of Google's acquisition of GIPS, who produced (and provided licenses for) the iLBC

Hello all, I've started implementing iLBC on some of the ATAs we have floating around clients' homes, but I'm coming against this error message with most of them: codec_ilbc.c: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (38)? The ATAs in question are various Grandstream models - the HT486 being the predominant one. Looking at the list archives, it's

hallo all, could sombody please help me, i dont know why nativ bridging is not working when i choose the ilbc codec, with speex it is working,?? iaxcomm (ilbc) ---> asterisk --> ( asterisk2 --> sip grandstream (alaw) ) \-----------------native bridge------------------/ 1. if i use on iaxcomm as default speex, nativ bridging between iaxcomm and my sip phone is working 2.

I want use iLBC and have following in mind, please help me is it possible ? ISDN <-----(ALAW)-----> * <-----(ALAW)-----> SNOM SIP??<-----(iLBC)----->?*?<-----(ALAW)----->?SNOM 1. ISDN incoming codec is ALAW SNOM codec should be ALAW (can't be iLBC because a lack of codec). 2. SIP incoming codec should be iLBC (snom is ALAW). 3. SIP outgoing codec should be iLBC /snom

At 6:27 AM +1130 on 5/15/04, Craig wrote: >Hi Brian, > >Out of interest, what do you used to measure data throughput and graph >it? > >I have been trying to find something to do realtime logging and graphing >of data like this. > >craig Realtime graphing is not supported, but this tool happens to be UNIX-based and parseable by other upstream systems (like RRDTool or

All, Has anybody else experienced garbled voice between a phone using alaw/ulaw and one using iLBC? I have a Nokia E series phone with a preference to use iLBC and this works fine in Asterisk 1.2. However, since moving to 1.4 - I get garbled voice on Inbound (g711->iLBC). Outbound voice seems fine (iLBC->g711) though. It's not a 20/30ms framing issue as the phone uses 30ms

I've been trying out the newest X-Lite (Build 1012) with iLBC and speex codecs. If I enable only iLBC _or_ SPX on X-Lite and call the echo-test on my asterisk server, the call connects, but I get no sound. If I enable only iLBC _and_ SPX, X-Lite indicated that it has connected with iLBC, and I hear a weird squawking. My sip.conf contains: allow=iLBC allow=SPEEX allow=gsm I've heard

i tried asterisk ilbc codec against kphone. when the call got connected, i started to immediately get these kind of message to the console: WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of 52 bytes long from RTP (50)? WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of

I already have ilbc installed on my system. The files are: /usr/include/ilbc/iLBC_decode.h /usr/include/ilbc/iLBC_define.h /usr/include/ilbc/iLBC_encode.h /usr/lib/libilbc.a /usr/lib/libilbc.la /usr/lib/libilbc.so -> libilbc.so.0.0.0 /usr/lib/libilbc.so.0 -> libilbc.so.0.0.0 /usr/lib/libilbc.so.0.0.0 However, if I do a "make" in asterisk-1.4.19, it will not detect that libilbc.a

Does anybody know if the updated licence on iLBC makes it safe to include in Asterisk when used in a commercial environment again? https://sites.google.com/site/webrtc/ilbc-freeware It seems to require that the Google iLBC licence document is on the box, but that otherwise it is free-to use by all in any way (BSD licence style). I believe that prior to that there was a requirement to register

Hello Folks, I noticed that all of the IaxClient based softphones with exception of Firefly only seem to have support for GSM but what about iLBC? The quality is excellent with iLBC even on a dialup connection! Meanwhile while the audio on GSM often sounds scratchy. Is anyone looking to implement iLBC in an IaxClient based softphone soon? Errol Biz4Web Solutions Limited

Hi! As the upstream of my DSL-connection is very slow, I'd like my sip-phones to use iLBC to connect to my *. My gateway provider only allows ulaw. Hence, I'd like to use the follwing setup: SIP-phone <--iLBC--> Asterisk <---ulaw----> PSTN-Gateway I get the following error: "Unable to find a codec translation path from ilbc to ulaw" Setup SIP-phone: disallow=all

I'm playing with codecs at the moment and have found some notices errors when x-lite/pro connects to asterisk with Speex or ILBC. Initially I was getting garbled sound, but after changing magic number for both codecs to 97 (as per http://www.voip-info.org/wiki-Asterisk%20phone%20xten%20xlite and http://bugs.digium.com/bug_view_page.php?bug_id=0000918) I was able to get normal voice. BUT,

This seems to have been reported before, but I've seen no resolution: http://lists.digium.com/pipermail/asterisk-dev/2003-August/001470.html http://lists.digium.com/pipermail/asterisk-users/2003-August/019091.html http://lists.digium.com/pipermail/asterisk-users/2003-October/024472.html When forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the Asterisk server) When forcing

Hi, I cannot get SIP channel working with folowing codec configuration: [sip] disallow=all allow=g723.1 ;I need this codec between sip phones (BT100) allow=ilbc ;Use this codec to others Calling between BT100 SIP phones is OK - asterisk makes native bridge (with g723.1) between them. When I'm calling from SIP to other channel (iax,zap,...), asterisk is not able to chose right codec

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