Greetings fellow listers, I have an application where I have approximately 300 files that I playback individually or in blocks to simulate "text-to-speech" in a "less mechanical" voice than normal Allison files provide. These files are presently in GSM format and sound pretty good when I play them on my computer speakers or on my in-house Polycom 501's over SIP connections. The "problem" I have is that the intended use of the application is going to be over SIP/DAHDI trunks that will connect to VM's over IAX trunks. What is your best suggestion for maintaining the quality of the audio as much as possible? Best Case presently - SIP phone in-house to IAX Worst Case presently - Cell phone calls Asterisk 1 on TDM400P which connects to VM Asterisk 2 via IAX. Asterisk version is 1.4.30 Thanks in Advance Danny Nicholas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100924/6662abec/attachment.htm
The best format would be in whatever format asterisk is sending the final audio out in. Even if you store it in the highest quality asterisk may have to transcode it on the fly so its best to store it in an already transcoded format to reduce the cpu load. For dahdi you would want to use the native .sln format. For sip use whatever coded you use over the sip connection. Danny Nicholas wrote:> Greetings fellow listers, > > I have an application where I have > approximately 300 files that I playback individually or in blocks to > simulate ?text-to-speech? in a ?less mechanical? voice than normal > Allison files provide. These files are presently in GSM format and > sound pretty good when I play them on my computer speakers or on my > in-house Polycom 501?s over SIP connections. The ?problem? I have is > that the intended use of the application is going to be over SIP/DAHDI > trunks that will connect to VM?s over IAX trunks. What is your best > suggestion for maintaining the quality of the audio as much as possible? > > > > Best Case presently ? SIP phone in-house to IAX > > Worst Case presently ? Cell phone calls Asterisk 1 on TDM400P which > connects to VM Asterisk 2 via IAX. > > > > Asterisk version is 1.4.30 > > > > Thanks in Advance > > Danny Nicholas >
Zeeshan Zakaria
2010-Sep-24 17:37 UTC
[asterisk-users] best format for playback/generation
If your sip provider supports gsm, then it is fine to send them your existing format, but I am sure by the time voice reaches an end user, it is transcoded at least once or twice again, so you can never guarantee what quality the end user is getting. I would stay with ulaw, as it has more chances to retain a better quailty even after a few transcodings, plus almost every sip provider will be able to receive it as it is and pass it on as received. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-24 1:02 PM, "Gareth Blades" <list-asterisk at skycomuk.com> wrote: The best format would be in whatever format asterisk is sending the final audio out in. Even if you store it in the highest quality asterisk may have to transcode it on the fly so its best to store it in an already transcoded format to reduce the cpu load. For dahdi you would want to use the native .sln format. For sip use whatever coded you use over the sip connection. Danny Nicholas wrote:> Greetings fellow listers, > > I have an ...-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100924/80b3413e/attachment.htm