similar to: iax client for symbian s60

Hello - We are building Speex 1.2rc1 on Symbian/S60 using the GNU csl-arm-2005Q1C tool chain provided in the S60 SDK. We build with the supplied config.h and define FIXED_POINT to 1. We found that the encoded audio is "garbled" when we build with GCC -O1 or -O2 optimization level with fixed point enabled. But if fixed point is enabled and no GCC optimization level is specified,

Quoting "Fabio Pietrosanti (naif)" <lists at infosecurity.ch>: > Speex it's too cpu expensive for general S60 usage, it would require a > lot of ASM optimization. Did a quick search and saw ARM CPUs with speeds above 100 MHz. That should actually be enough for Speex, at least for narrowband. > If you are using CSD, like for a secure telephony solution >

It would be very nice to know about some good success without too much cpu issues on Nokia devices, i tried pjsip.org project on S60 FP1 E65 +200mhz ARM cpu with speex but don't had much luck in using it without having 100% cpu usage :( Please let us to know about your testing E65 CPU: http://www.nokia-tuning.net/index.php?s=processor Fabio Pietrosanti Jordan Dimov wrote: > Thank

Is speex at this stage a good codec to use for real-time voice transmission over a direct CSD connection between two phones? Does anyone have any experience with using Speex on Nokia phones? If speex is not a good solution for this task, what would you recommend? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, If I was going to post this as an iPhone-only SIP client, I'd expect loud booing and hissing, but Media5 mobile SIP client is available for the Symbian S60 platform, too, or will be shortly. Interested? To join us and hear about Media5 form Pascal Dore, see http://vuc.me Speaking of mobile, if any of you know the people who are developing the SIP client for Android, we'd love to

Thank you, guys. Most smartphones in production today run at ~300MHz and above. So this shouldn't be a problem. I'll try to compile speex for Symbian (I see it's been ported, but the Symbian port is somewhat out of date?) and I'll run it on a pair of Nokia N76 phones. I'll let you know how it works out. If someone has any further advice in this direction, I would deeply

Fabio Pietrosanti (naif) wrote: > i am not criticizing the technology itself, but the fact that from the > point of view of the implementator that need to make a technological > choice while selecting a codec to be used for a mobile multimedia > application, speex does not provide affordable informations on > "supported platform".

Hi, I'm trying to compile speex1.2-rc1 for Symbian 3th edition. I get some errors: ***Invoking bldmake command perl.exe -S bldmake.pl bldfiles winscw \Users\Maurizio\University\ProgettoInterdisciplinare\speex-1.2rc1\speex-1.2rc1\symbian\BLD.INF FATAL ERROR(S): \Users\Maurizio\University\ProgettoInterdisciplinare\speex-1.2rc1\speex-1.2rc1\symbian\BLD.INF(37) : Exported source file

Hi all, i am not criticizing the technology itself, but the fact that from the point of view of the implementator that need to make a technological choice while selecting a codec to be used for a mobile multimedia application, speex does not provide affordable informations on "supported platform". Without arranging a test bed with a lot of devices and spending a lot of money i

hi all, I can claim that our hotfoon service has run millions of mobile voip minutes using speex on mobile and desktops. it works! earlier you needed more powerful cpus (until the integer port), however now the cpus are much faster and the code complexity is much lower. my only regret is of not being able to write some demo apps for speex on win mob. I will surely deliver it in june to settle this

Hi anyone know where i can find all internatinal numbering plan in csv and for free or small price ? thanks Jpc

Hi all, I've released another free app for the iPhone and iPod touch - this one lets you read the Daily Asterisk News. Hope you enjoy it :D http://www.venturevoip.com/news.php?rssid=2371 -- Cheers, Matt Riddell Managing Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP

G'day all, I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone number and then the three of them would be in a conference. From there they should be able to bring in other people as well. This seems to be what the Asterisk n-way call HOWTO

hello, please any IAX2 ActiveX control that wrap libiax2 or libiaxclient? i want to develope my softphone in delphi thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4345 (20090818) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . Regards H.Motamedi --------------

I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. Let me know. Thanks. -------------- next

Hi, I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: max_connection=1000 wait_timeout=60 query_cache_type=1 query_cache_limit=4M query_cache_size=512M interactive_timeout=120

I am running Asterisk and using Answer machine detection with call files on a virtual Vcloud server running Centos 5.3 and LAMP. I am finding that AMD is only detecting HUMAN or MACHINE for about 30% of the calls (I sent over 50,000 outbound calls last week, and 70% said NOTSURE). I have a suspicion that the problem may be due to the timing source on virtual server when its under load delivering

Hi, I was looking round on the Internet and saw there was no definitive list of free applications available for use with Asterisk, so I thought I'd compile a list for you all. If there's anything that you know of that is actively maintained but not in the list below, let me know (bear in mind I'm not including distros or Asterisk packagings in this list). Hopefully there are a few

Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090910/af1f9656/attachment.htm