[asterisk-ss7] Chan_ss7 telco interconnection requirement

[Anton]

Does someone has an equipment to measure that values over an 
Asterisk solution?

On 22 November 2006 20:06, Sam Njenga wrote:
> Below are the questions. Your detailed answers will be
> highly appreciated.
>
> 1.       Confirm the offered system meets the guaranteed
> maximum value of voice distortion and specify the offered
> solution's design distortion values.
>
>
>
> 2.       Confirm the offered system has a guaranteed
> maximum value of loudness and specify the offered
> solution's design loudness criterion.
>
>
>
> 3.       Confirm the maximum guaranteed jitter and
> explain how it compensates for, or minimize jitter in
> their system (including the design criteria for size of
> the jitter-buffers).
>
>
>
> 4.       Confirm the maximum guaranteed packet loss-rate.
> Confirm this rate and the adopted strategy for
> compensating for, or minimising packet loss in the
> system.
>
>
>
> 5.       Confirm that the system employs
> industry-standard or proprietary measures for packet
> recovery.
>
>
>
> 6.       Confirm the offered system has a guaranteed
> maximum value of system-additive noise and specify the
> offered solution's design level of additive noise.
>
>
>
> 7.       Confirm the offered system has a guaranteed
> maximum value of fade-distortion and specify the offered
> solution's design fade level.
>
>
>
> 8.       Confirm the offered system has a guaranteed
> maximum value of cross-talk and  shall specify the
> offered solution's design cross-talk limit.
>
>
>
> 9.       Confirm the offered solution has a guaranteed,
> well-defined design voice quality score against an
> objective industry-standard clarity assessment system
> (such as ITU-T P.561/P.562). Either intrusive or
> non-intrusive measurement schemes shall be used and
> specify the test strategy including the test
> frequency-domain span.
>
>
>
> 10.    Confirm the equipment achieves silence suppression
> and injects comfort noise into the circuit, without loss
> of the side-tone and  specify the offered solution's
> design voice-suppression depth
>
>
>
> 11.    Confirm the offered solution has a guaranteed
> maximum latency. The contributions to the latency shall
> be disclosed.
>
>
>
> 12.    Confirm the offered solution has echo-canceller
> meeting or exceeding performance in ITU-G.165/G.168, and
> able to support ITU-T G.168-2002. An objective measure of
> PACE shall be specified.
>
>
>
>
>
> 13.    Confirm the offered system compensates for
> front-end clipping and holdover suffered by a voice
> signal input. The implemented mitigation scheme shall be
> specified.
>
>
>
> 14.    Confirm the offered solution has bandwidth gain.
> The techniques used to achieve the gain and the
> contribution from each technique shall be specified.
>
>
>
> 15.    Confirm the offered solution's optimal
> environmental condition metrics. The optimal ambient
> metrics shall be disclosed (including but not limited to
> optimal temperature, altitude, air pressure, humidity,
> noise and shock resistance).
>
>
>
> 16.    Confirm the offered system's electromagnetic power
> immunity from and emission into the environment and 
> shall clearly state the compatibility certification for
> the proposed system.
>
> On 11/22/06, Anton <anton.vazir at gmail.com> wrote:
> > Can you post that questions here?
> >
> > On 22 November 2006 19:04, Sam Njenga wrote:
> > > Hi
> > > I am interconnecting to a telco provider using
> > > chan_ss7. I have opted to use asterisk for this. I am
> > > quite conversant with asterisk configuration and
> > > little knowledge about ss7. I have done back-2back
> > > tests between two asterisk boxes and calls pass
> > > perfectly. The local telco has requested me to fill a
> > > technical interconnection questionare about my
> > > system. Some of the questions require detailed
> > > explanations.
> > >
> > > Can someone with SS7 knowledge assist me on this ?
> > >
> > > Sam
> >
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