search for: ilbc

Hi, I try to set up planet VIP-050 with asterisk. Everything works fine instead of the call transfer. When I press # console says something like this: >Oct 5 11:11:20 DEBUG[25104]: chan_sip.c:2222 sip_rtp_read: Oooh, format changed >to 1024 >Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc >frame that isn't a multipleof 50 bytes long from RTP (4)? >Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc >frame that isn't a multipleof 50 bytes long from RTP (4)? >Oct 5 11:11:20 WARNING[25104]: codec_...

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: disallo...

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 codec. We've determined that...

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 codec. We've determined that...

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, L...

I'm not an expert either, but I see people choosing iLBC over speex all the time with asterisk; partly it's because they have more market share in hardphones, and partly it's because of marketing and such. (another reason is that iLBC source is included in asterisk, and speex is only compiled in if you have the speex development stuff...

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 inco...

Hi kind developers, I need select soon the best freeware VOIP codec, I see that all competitors are using iLBC because of the separate packets management. How speex behave in case of packets drop? Why other choice all iLBC? Thank you for any kind answer. Best regards. ------------------------------------- Roberto Della Pasqua Http: www.dellapasqua.com Email/Msn: roberto@dellapasqua.com Aol/Yahoo: DellaP...

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,...

I have around 20-30 softphones behind NAT .. My sip.conf has nat=yes and they all are able to register and make calls with no problem . My voip carrier supports gsm as well as ilbc .. Server takes calls from sip phones , does call recording in between and forwards to voip carrier . My problem is that half of my softphones use ilbc and rest use gsm and my provider supports both gsm as well as ilbc . Now when i put allow=gsm&ilbc in my voip carrier's extension then it...

Hi, Is there any comparison made between Speex and iLBC free codec? How would they compare in terms of quality, bitrate and CPU utilization? Thanks, Joe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20050610/b79a3f46/attachment.htm

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 p...

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 i...

...OTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 15:50:27 NOTICE[1192491824]: channel.c:1508 ast_set_read_format: Unable to find a path from G726 to SLINR Jul 9 15:50:27 NOTICE[1192491824]: channel.c:1478 ast_set_write_format: Unable to find a path from ILBC to G726 Jul 9 15:50:27 WARNING[1192491824]: codec_ilbc.c:141 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (80)? Jul 9 15:50:27 WARNING[1192491824]: chan_sip.c:1333 sip_write: Asked to transmit frame type 1024, while native formats is 16 (read/write =...

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 tha...

.../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 iLBC, there is some very garbled noise, but nothing intelligible. Sniffing the packets, I can see that X-Lite & Asterisk have chosen differing 'Payload type' numbers: X-Lite: a=rtpmap:97 speex/8000 a=rtpmap:98 iLBC/8000 Asterisk: a=rtpmap:97 iLBC/8000 a=rtpmap:110 SPEEX/8000 According to...

...Nothing can reinvite, this path is forced for the entire call. +--------+ +--------+ +--------+ | gs | <-----> | sip0 | <-----> | sip1 | +--------+ +--------+ +--------+ I would like the RDP traffic between the gs and sip0 to be encoded using ILBC and the traffic between sip0 and sip1 to use G.711. I can force the gs/sip0 path to ILBC be allowing only that codec in the gs's sip config, however, even when I specify in the sip config that sip1 can only use ulaw, it uses ILBC, as observed from a `sip show peers`. sip1 allows both ILBC and...

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

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. if i use ilb...