Hi Christopher, been thinking about this for a little bit and was wondering if instead of going on a chase around the net to find how to do this, would you be able to share details of how you have your TTS / MRCP Cluster setup? Is this in the Wiki somewhere or did you figure this out on your own? How well does this perform? I'm assuming if I can manage to compile FS or Asterisk (depending on which one I go with) on Solaris 10, then I could not worry about TTS compiling on Solaris as I can just build my MRCP cluster like yours on Linux/Windows etc and just do what you're doing? Any help would be HIGHLY HIGHLY appreciated. Thank you in advance RR On Thu, Nov 11, 2010 at 8:57 AM, Christopher Rienzo <cmrienzo@gmail.com>wrote:
I can't answer about FreeSWITCH on solaris, but I can talk about TTS. In my setup, I have a cluster of MRCP servers. The FreeSWITCH boxes then use mod_unimrcp to make TTS requests to that cluster. This has the following benefits:
1. TTS licenses can be shared among any FreeSWITCH box. This is good because licenses are expensive. 2. There are more TTS vendors that support MRCP than there are TTS modules in FreeSWITCH- though a new module could be written if you are a C developer.
Others use mod_cepstral and mod_tts_commandline, so those options can be explored too.
On Thu, Nov 11, 2010 at 6:32 AM, RR <ranjtech@gmail.com> wrote:
Hello All,
I have a whole bunch of questions regarding the TTS functionality in FreeSWITCH and Just Generally about Freeswitch on a Sparc based Solaris Platform and then the combo of the two, whether FreeSWITCH can be built on a Sparc based Solaris platform when compiled with a "reasonably" good TTS module be it Cepstral, FLite OR NeoSpeech etc?
1) Can anyone reflect on any experience they might have had in terms of the performance difference between TTS on Asterisk (let's say Asterisk 1.8) and FreeSWITCH? Is one better than the other? 2) Can One or the other or both be compiled successfully on a Sparc based Solaris platform? 3) Would it benefit if I was to use a dedicated box for TTS and throw calls to that box for TTS processing and send back media the way it came after the sound files are generated from the TTS-process? 4) Please note that the quality of the Translation doesn't need to be impeccable or mimic human speech. It just needs to be understandable enough esp. over mobile phones. The more complex part about it is, that the user directory can/will have names from all sorts of ethnicities/nationalities which may mess up a lot of these TTS engines. So what is the best approach there in this mixed environment 5) Does the OpenMRCP play any part in any of this? Would I benefit with that?
I think this is the crux of my concerns. Any ideas and/or suggestions to kick off the discussion would be very very much appreciated.
Kind Regards, RR
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