We are using Polycom phones and I don't think they support speex or ILBC (at least not all the models).
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten Sent: Tuesday, April 20, 2010 3:57 PM To: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] g.729 FYI I just confirmed using the X-Lite Client that FreeSWITCH supports both the narrowband and wideband speex codecs. From what I understand it's lower bandwidth than G.729: http://lists.xiph.org/pipermail/speex-dev/2006-May/004453.html I couldn't get iLBC to work with FreeSWITCH/Polycom/xLite so I gave up on it. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 4/20/2010 1:29 PM, Josh Patten wrote: Does your ITSP support speex or iLBC? If so, consider the following: in /etc/sipxpbx/freeswitch/conf/sip_profiles/sipX_profile.xml: <param name="codec-prefs" value="G722,p...@20i,p...@20i,speex,L16"/> speex is supported by FreeSWITCH and it's a very robust low bandwidth codec albeit at some CPU cost. It's also already preconfigured in sipX's implementation on FreeSWITCH. Adding iLBC should be as simple as changing that line and possibly a few others in other FreeSWITCH config files by adding i...@30i to that list. Perhaps you could do G.729 for phone-to-phone calls and then let FreeSWITCH do speex? Am I missing a reason why this would not be possible? Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 4/20/2010 1:13 PM, Michael Scheidell wrote: I really ONLY care about AA/Voicemail/Conferencing (since everything else seems to work). I am assuming the only time I will need it is in freeswitch, caller using G.729, hits AA/vmail or conference. I get 4 licenses, I am assuming 5th concurrent caller (using G.729) will probably not hear anything. in the case of conferencing, oh well. in the case of AA or vmail, maybe he will call back. maybe 10 licenses ($100) isn't all that much to spend if it works. I thought I did turn off everything in voip.ms except G729 and the caller can't reach AA or Vmail if the user doesn't answer. (you on 4.2.0 yet?) On 4/20/10 2:08 PM, Eric Varsanyi wrote: I may be misunderstanding your configuration, but it seems like if you don't care about AA/Voicemail/Conferencing then the freeswitch license isn't going to help (or hurt) you at all. If your remote callers would be better off with a different codec you could try to limit the codecs their endpoint s/w or h/w will allow (ie: don't advertise g711 at all, just uLaw and g729). If your remote users are on POTS then your issue is between you and your ITSP and you could limit your ITSP configuration to the codecs you like (for instance with voip.ms I can pick which codecs it will accept, I turned off g711 and gsm so now it always negotiates uLaw which is pretty good; I wish they had g722 as an option too though). -- Michael Scheidell, CTO Phone: 561-999-5000, x 1259 > | SECNAP Network Security Corporation . Certified SNORT Integrator . 2008-9 Hot Company Award Winner, World Executive Alliance . Five-Star Partner Program 2009, VARBusiness . Best Anti-Spam Product 2008, Network Products Guide . King of Spam Filters, SC Magazine 2008 _____ This email has been scanned and certified safe by SpammerTrapR. For Information please see http://www.secnap.com/products/spammertrap/ _____ _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/ _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
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