search for: trancode

Good day all I want to know what the best codec is to use for asteris for VOIP We have two towns connected with a 64k line that's going to do VOIP with astersik.At the moment with the default installation the quality is bad and the bandwith is high. Is this even a codec problem Pleas help ALtus

I've got a question about codec_dahdi witrh a system running Asterisk 1.6.0.6 and DAHDI 2.1.0.4 with a TE410P card. The system is used primary to route calls between different PRI connections, so no transcoding between codecs is happening as far as I am aware. 1) How can I use codec_dahdi? Would it be useful when passing a call from one dahdi channel to another dahdi channel? 2) I'm

...a good idea? On Fri, 04 Mar 2016 14:48:41 +0100, you wrote: >Great tool to do this: liquidsoap > >Keep in mind that transcoding degrades the quality of tour stream dramaticly. You can avoid this by feeding liquidsoap or stream transcoder with a highquality or even transparant stream and trancode this to the different streaming formats you like. I used to do this by feeding a flac stream to liquidsoap and transcode this to 5 different stream formats i needed. But you might be fine if you feed your transcoder with 320 kbps AAC and transcoder this to lower formats. Maybe experiment with a hig...

All the broadcasters on the server which I support deliver their content in MP3 format. Recently, there's been interest in supplying a second AAC stream at half the bandwidth but with the same audio quality (64kbps AAC versus 128kbps MP3) like TuneInRadio does for delivering their content regardless of the source. I've thought of using a third-party product called Stream Transcoder, but am

On Sat, 2003-05-31 at 10:29, Stefan Neufeind wrote: > I have mp3s as input and can't convert them all. And since my > listeners will ONLY use mp3-streaming I don't want the overhead and > quality-loss of converting mp3 to ogg and later stream-transcode them > from ogg back to mp3. That's not worth it. > So why can't support for mp3 be added to ices2? From a

On 31 May 2003 at 12:49, Karl Heyes wrote: > On Sat, 2003-05-31 at 10:29, Stefan Neufeind wrote: > > I have mp3s as input and can't convert them all. And since my > > listeners will ONLY use mp3-streaming I don't want the overhead and > > quality-loss of converting mp3 to ogg and later stream-transcode > > them from ogg back to mp3. That's not worth it.

...ages; or any other signaling gateway > e.g. socketio > 2. Capable to generate "answer-sdp" and return backing using same > signaling gateway. (make sure that ICE are gathered & included in the > anwer-sdp) > 3. For WebRTC, media server MUST support DTSL/SRTP to capture, trancode > and forward encrypted RTP/RTCP pakcets > > I don't know if icecast2 has such functionalities built-in. It currently > accepts PUT requests only on HTTP. > > ?You can install STUN server side by side with icecast2 on a separate > port.? > > ?I think a gateway (a midd...

Great tool to do this: liquidsoap Keep in mind that transcoding degrades the quality of tour stream dramaticly. You can avoid this by feeding liquidsoap or stream transcoder with a highquality or even transparant stream and trancode this to the different streaming formats you like. I used to do this by feeding a flac stream to liquidsoap and transcode this to 5 different stream formats i needed. But you might be fine if you feed your transcoder with 320 kbps AAC and transcoder this to lower formats. Maybe experiment with a hig...

Hi, I would like to improve my understanding of T.38. 1. What T38FAX_VERSION_0 or T38FAX_VERSION_1 in chan_sip.c means ? voip-info.org implies one has to change values in chan_sip.c to make it work. Shall I set T38FAX_VERSION_1 or leave T38FAX_VERSION_0 in global_t38_capability ? Source code says "This is default: NO MMR and JBIG trancoding, NO fill bit removal, transferredTCF TCF, UDP FEC,

Hello! I use asterisk with TE420 as PRI switch for two channels : ;panasonic uplink group=3 context=panasuplink ; relaxdtmf=yes ; immediate=yes rxgain=0.0 txgain=0.0 mohsuggest=default jbenable = no ; jbenable = yes ; jbmaxsize = 200 ; display_send=name_initial display_send=name

Need some help understanding codec preferences: I have 2 asterisk servers. Server 1 sends calls to the PSTN and has allow=g729 allow=gsm and allow=ulaw in iax.conf Server 2 receives calls and routes them to server 1. It has the same allow lines. We receive calls from a phone co and route them via server 2 to server 1. The calls originate in g729 and everything works fine. Now I want to take

Hi i'm trying to connect to the PSTN in the following way sip ATA -> * -> gnugk -> Cisco AS5300 -> PSTN I'm using asterisk CVS-HEAD-06/01/05-14:33:15 running over RH EL3 Asterisk-Oh323 0.7.2 pre1 Open H323 v1.13.5 pwlib v1.6.6 and I'm having a lot of trouble, gnugk and * both have public ips and are not behind any type of firewall, the sip ATA is behind a firewall and

> Best wishes > Stefan (on second thought ) :: my last post assumed that you had .OGG files to begin with .. depending on the size of your library, you may want to convert your mp3s to ogg . i ended up doing this , and it made things much simpler on the web end . also ; check out dir2ogg .. echo "dir2ogg written by Darren Kirby; linuxbot@shaw.ca; www.badcomputer.no-ip.com/"

Hi all - I am using asterisk 13.27.0 with Linphone. I turned off all codes on linphone except the one I want to try. For example: opus and speex (so only one enabled at a time). Then did this same on asterisk for the linphone extension. disallow=all allow=speex (for example). Then I place my call and the call fails. if I enable something like gsm, ulaw, alaw the call works fine. Why does the