[asterisk-users] Teliax Quality of Service

Anthony Francis anthonyf at rockynet.com
Fri Aug 3 14:12:11 CDT 2007


Haudy Kazemi wrote:
> On Aug 2 2007, John Meksavan wrote:
>
>   
>> Asterisk Users,
>>
>>  I recently ran into some problems with the quality of service with 
>> Teliax.
>>  This occurred on August 1, 2007 with a dropped outbound call, audio 
>> quality isse on the callee side- not hearing me well on callee side, and 
>> sending DTMF tones (configured for RFC2833). Am I the only Teliax 
>> customer having this problem?
>>
>>  It seems like when I am ready to go live with my Asterisk PBX System, I 
>> run into quality of service issues with the SIP provider.  Who should I go 
>> with that would guarantee me quality service just like an analog line?
>>     
>
> VoIP is susceptible to packet delivery problems anywhere between your PBX 
> and your SIP provider's PRI lines/termination point. If you have direct SIP 
> PBX to SIP PBX calls, then your problems can be anywhere on the Internet 
> path between the sites. The only workaround that I know of is having your 
> ISP be your SIP provider, so that your SIP packets only cross your ISP's 
> own network to its termination point, and do not cross the public Internet. 
> This way QoS can work from your office to your ISP's office to make sure 
> you maintain reliability.
>
> I have not personally used iTEL-ip's 'iTEL Voice Service', but others have 
> said, as do their own notes that their network QoS is effective at 
> maintaining call quality. When I contacted them, their pricing for a 'QoS 
> private IP backbone for voice and data' was $618/month for a full 1.5mbps 
> T1. Then SIP trunks (#11-24) were anywhere from $10-12 per month depending 
> on contract length. Per minute rates were $.03.
>
> When I ran the numbers, it appeared that a regular full T1 + a regular full 
> PRI would be only slightly more. A major tradeoff comes in the physical 
> location flexibility you get with SIP over traditional phone lines in the 
> case you need to move an office (although physically moving the phones to a 
> non iTEL-ip data line would mean you're not getting their Qos).
>
> iTEL-ip's 'iTEL Voice Service' 
> http://www.itelconnect.com/default.aspx?type=t&section=iTEL-ipVoiceService&selection=16
>
> http://wiki.pbxnsip.com/index.php/ITEL-ip
>
> -hk
>
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On the original problem of missed DTMF set dtmfmode=info in your sip.conf.

Anthony



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