search for: isac

...capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs) Modular ISDN Stack core $Revision: 1.37 $ mISDNd: kernel daemon started (current:ca4d8680) mISDNd: test event done ISDN L1 driver version 1.18 ISDN L2 driver version 1.31 mISDN: DSS1 Rev. 1.42 mISDN Capi 2.0 driver file version 1.20 ISAC module $Revision: 1.17 $ mISDN_dsp: Audio DSP Rev. 1.24 (debug=0x0) EchoCancellor MG2 dtmftreshold(100) mISDN_dsp: DSP clocks every 64 samples. This equals 8 jiffies. DTMF modul version 1.16 Traverse Tech. NETjet-S driver, revision 1.6 nj_probe(mISDN): found adapter NETJet S at 0000:00:0b.0 NETJet...

I had Asterisk 1.6.2.2 running fine with a mISDN using a HFC-S based card. I installed my TDM400P into the PC, it's really slow to boot now, when it finally does I gets stuck in a loop of reporting "isac xdu no tx_busy". Anyone able to assist? Thanks in advance! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100219/add17ccf/attachment.htm

Besides the codecs that * supports. Is there any ISAC implementation for asterisk available? This is to be used mainly with softphones, i haven't seen any hardphones that support this codec. Thanks, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

...OAD, assuming '0' isdn_net: call from 0 -> 0 xxxxxxxx ignored where xxxxxxxx is my local phone number without area code I turned on the maximum isdn debugging, and here is the output when I dial in, at this time, asterisk was not running... 29:42.76 Card1 tiger: i1 2 80 29:42.76 Card1 ISAC interrupt 80 29:42.76 Card1 isac_empty_fifo cnt 4 02 81 01 3B 29:42.76 Card1 <- PH_DATA: RR[1](nr 29)C (sapi 0, tei 64) HEX: 02 81 01 3B 29:42.76 frame network->user SFrame with tei 64 29:42.78 Card1 tiger: i1 2 80 29:42.78 Card1 ISAC interrupt 80 29:42.78 Card1 isac_empty_fifo cnt 4 00 81 01...

...alled Asterisk along with chan_capi from apt-get. I installed mISDN from the CVS of isdn4linux.de. It is : - Asterisk 1.0.7 with bristuff - chan_capi 0.3.5 When I load the whole modules lot, I get the following in dmesg: Modular ISDN Stack core $Revision: 1.25 $ mISDNd: kernel daemon started ISAC module $Revision: 1.16 $ mISDNd: test event done CAPI Subsystem Rev 1.1.2.8 capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs) ISDN L1 driver version 1.11 ISDN L2 driver version 1.20 mISDN: DSS1 Rev. 1.30 mISDN Capi 2.0 driver file version 1.14 X25 DTE modul version 1.8 AVM Fritz PCI...

...atus(2) after 50000us init_card: entered inithfcpci: entered HFC PCI: IRQ 18 count 19 HFC card c8554800 dch c855487c bch1 c8554a18 bch2 c8554bb4 mISDN: HFC-PCI driver Rev. 1.49 HFC-PCI: No more PCI cards found HFC 1 cards installed /sbin/modprobe --ignore-install avmfritz protocol=0x2 layermask=0xf ISAC module $Revision: 1.18 $ AVM Fritz PCI/PnP driver Rev. 1.43 PCI: Enabling device 0000:00:00.0 (0000 -> 0003) mISDN_fcpcipnp: found adapter Fritz!Card PCI at 0000:00:00.0 fritz card c8554800 dch c85548a0 bch1 c8554a3c bch2 c8554bd8 AVM PCI: stat 0x3020a AVM PCI: Class A Rev 2 AVM PnP: HDLC versio...

What's the impact on encoded speech quality (per given bitrate) when the encoder cpu complexity is reduced all the way down for Opus? Rather, how big is the impact? Secondly, can someone comment on wideband speech quality comparison between Opus and iSAC with and without the cpu complexity of Opus turned all the way down? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20150113/685c1962/attachment.htm

...level ????a=sendrecv ????a=mid:audio ????a=rtcp-mux ????a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:haR/UikskQr/AIrry5udqINI1hYfc5zY2I6jrkKT ????a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:waQfKIHI9UyjPVI0vrcUREDbSVZdtfCtRQK71/Ks ????a=rtpmap:111 opus/48000/2 ????a=fmtp:111 minptime=10 ????a=rtpmap:103 ISAC/16000 ????a=rtpmap:104 ISAC/32000 ????a=rtpmap:0 PCMU/8000 ????a=rtpmap:8 PCMA/8000 ????a=rtpmap:107 CN/48000 ????a=rtpmap:106 CN/32000 ????a=rtpmap:105 CN/16000 ????a=rtpmap:13 CN/8000 ????a=rtpmap:126 telephone-event/8000 ????a=maxptime:60 ????a=ssrc:2121187131 cname:RXEEP3aaYIHOpxRX ????a=ssrc:2...

...:5E:B0:E7:E3:F9:D8:C2:F4:60:2A:D0:76:46:F8:3A:97:01:C9:0C:5A:F7:B8:7D:C1:43 a=setup:actpass a=mid:audio // a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level // a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time // a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2179369454 cname:SvzCJjIAujxHGm9P a=ssrc:2179369454 msid:Kqg5QpXyqNeviT8qxUIRi8QNUaV7mUnFiDIF add6e533-c83d-...

One of my user is praising Skype!!! I cannot figure out anymore what I can improve! This users sip show peers is jumping from 65 msec to 1800 all the time. Of course his voice quality is like a morse code with dashes or dots of connection time. The next minute he calls me via Skype and it works fine !!!! What indicates that there is no fault on his Internet connection!!! He is using his

...p:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:dYMEPP1zoNS/W70Ln6cnBCtHXDCq6ciLZmHDHdFj a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Gr23SpFGDiukOKyrrfAauWssQ+3pYjD0jwkK9hOo a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2003565451 cname:0Cqf6EiGG5oFoWF5 a=ssrc:2003565451 msid:dXVhxyOSxULu3iClZayhTeEBzH2voboiJJ28 8ae66e18-c0ba...

> I am asking this because it is believed that Skype is using some iLBC and > iSAC since GlobalIPSound listed Skype as a partner. I think (from what I've heard) that's what Skype uses. I have no idea how iSac sounds because it's proprietary and I've never used Skype. Jean-Marc > > Thanks, > Joe > > -----Original Message----- > From: Jean-M...

...he same can be done with textpad and adjustments in preferences/file also changes owner when a file is changed and saved. greetings jens Kramer -----Urspr?ngliche Nachricht----- Von: Willem Jaap Zwart [mailto:W.J.Zwart@NescioLudens.nl] Gesendet: Freitag, 8. April 2005 16:55 An: Kramer Jens ZFF ISAC Cc: samba@samba.org Betreff: Re: [Samba] Samba 3.10 and higher Hi We noticed this as well. This is because ultraedit effectively moves the original file to the bak file first and then create a NEW file (with indeed the rights of the current user). Because of the group writable bit this is compl...

...chs.More streams in, worse voice got. Then we write test code for opus-codec which encode a .pcm file simultaneously. and get error. we also write decoder test code. error too. We tested v1.0.2 and 1.0.3, Our codec is a .so file, our main program is written in erlang. we used other codecs such as isac/ilbc/... they are OK. you may try lwork.hk:8086, it is a webRTC demo. so, anybody noticed this? It seems to be a re-entry bug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20131018/3aadc00e/attachment.htm

...DP 2122300926 192.168.0.176 19505 a=candidate:1 2 UDP 2122285310 169.254.2.2 19505 a=candidate:2 2 UDP 2122285054 192.168.238.1 19505 a=candidate:3 2 UDP 2122284798 192.168.111.1 19505 a=candidate:5 2 UDP 1694482430 193.227.186.146 19505 a=candidate:6 2 UDP 16744446 87.236.144.70 41344 a=rtpmap:103 ISAC/16000 a=fmtp:18 annexb=no a=rtpmap:102 iLBC/8000 a=rtpmap:97 IPCMWB/16000 a=rtpmap:119 ISACLC/16000 a=rtpmap:117 red/8000 a=rtpmap:100 EG711U/8000 a=rtpmap:101 EG711A/8000 a=rtpmap:105 CN/16000 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=sendrecv <-------------> --- (16 headers 33 li...

...10:45:24] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level [May 10 10:45:24] a=sendrecv [May 10 10:45:24] a=rtcp-mux [May 10 10:45:24] a=rtpmap:111 opus/48000/2 [May 10 10:45:24] a=rtcp-fb:111 transport-cc [May 10 10:45:24] a=fmtp:111 minptime=10;useinbandfec=1 [May 10 10:45:24] a=rtpmap:103 ISAC/16000 [May 10 10:45:24] a=rtpmap:104 ISAC/32000 [May 10 10:45:24] a=rtpmap:9 G722/8000 [May 10 10:45:24] a=rtpmap:0 PCMU/8000 [May 10 10:45:24] a=rtpmap:8 PCMA/8000 [May 10 10:45:24] a=rtpmap:106 CN/32000 [May 10 10:45:24] a=rtpmap:105 CN/16000 [May 10 10:45:24] a=rtpmap:13 CN/8000 [May 10 10:45:24...

Thomas, Thank you for your interest in this, you description is as accurate as I can see. > From my perspective your challenges will be to get the containers right. > WebM for audio+video > Ogg for audio > > Also (I'm not that familiar with webRTC) you might need to reencode > to Opus and VP8 in some cases? here is the great news

...e codecs are already supported in webrtc :) > > Yes, I'm aware of that, but I don't understand webRTC well enough to > know if you can get away with a participant that only advertizes > support for the above and doesn't support: > http://www.webrtc.org/faq#TOC-What-is-the-iSAC-audio-codec- > http://www.webrtc.org/faq#TOC-What-is-the-iLBC-audio-codec- > and whatever else might still appear. > > That is nothing to worry about for a first proof of concept though. > Start out with VP8 and Opus, worry about the rest later. > >> It seems to me the bigge...

...4:AF:21:E2:A1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=sendrecv a=rtcp-mux a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:3696151487 cname:jXfPZ33h32Mx9liw a=ssrc:3696151487 msid:cC3clldcCIxZyWm8eHpKdycUakfANC...

...ext:ssrc-audio-level [Aug 9 22:15:50] a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time [Aug 9 22:15:50] a=sendrecv [Aug 9 22:15:50] a=rtcp-mux [Aug 9 22:15:50] a=rtpmap:111 opus/48000/2 [Aug 9 22:15:50] a=fmtp:111 minptime=10; useinbandfec=1 [Aug 9 22:15:50] a=rtpmap:103 ISAC/16000 [Aug 9 22:15:50] a=rtpmap:104 ISAC/32000 [Aug 9 22:15:50] a=rtpmap:9 G722/8000 [Aug 9 22:15:50] a=rtpmap:0 PCMU/8000 [Aug 9 22:15:50] a=rtpmap:8 PCMA/8000 [Aug 9 22:15:50] a=rtpmap:106 CN/32000 [Aug 9 22:15:50] a=rtpmap:105 CN/16000 [Aug 9 22:15:50] a=rtpmap:13 CN/8000 [Aug 9 22:15:50...