howto test/measure quality of a VoIP service?

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Foiler

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#237973 27-Jun-2018 12:10

Can anyone assist with methods to measure the quality of a VoIP service? I am interested in measuring total delay for voice in each direction or the total turnaround .. also how to quantify other impairments eg distortion, noise, audio dropouts/gaps ..

I have a customer whose voice quality is very poor, and a delay which seems to approach 1 sec. I am thinking of recommending they change to a 2Talk solution (which they are already trialing), but it would be good to have some hard data.

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#2044705 27-Jun-2018 12:41

Various ways of doing this - is there a SIP PBX involved on premise or just handsets connecting straight back to the VoIP provider? Ways to measure the data will depend on the handsets in use or PBX

Foiler

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#2044709 27-Jun-2018 12:47

jnimmo:

Various ways of doing this - is there a SIP PBX involved on premise or just handsets connecting straight back to the VoIP provider? Ways to measure the data will depend on the handsets in use or PBX

No PBX, and two different desk phones in use .. Cisco SPA504G and Cisco SPA525G

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#2044711 27-Jun-2018 12:49

If you have a Cisco router (CPE) in your network, can could look at something like Cisco IPSLA.

That will produce things like MOS scores etc, but needs a Cisco router on the other end to reflect off.

Sometimes your provider or PABX will supply stats like MOS score per call etc.

sbiddle

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#2044727 27-Jun-2018 13:43

This question really falls into the "if you have to ask it's probably too hard to do" category.

Cisco SPA phones will all dump full SIP RTP stats in the BYE message if configured. This shows total packets, lost RX / TX packets and jitter. Your challenge is to capture these - obviously Wireshark using a mirrored port would be the easiest.

The phones also display stats in the web interface during a call.

If you have 1 second delay you have much bigger issues. Where is the SIP proxy they're connecting to? On the other side of the world? What type of connection are you on?

Foiler

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#2044964 27-Jun-2018 19:56

Customer is connected by Vodafone fibre 50/10 to Voyager's VoIP service .. just to clarify, the delay in one direction is about 0.5 sec on a mobile phone call.

You would think there might be tools available to compare the quality of VoIP from different providers? Or is sophisticated test equipment required for this? Something along the lines of an audio ping to a loop-back handset and then measure the round trip delay and distortion plus noise would be a good start.

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#2044967 27-Jun-2018 20:03

Foiler:

You would think there might be tools available to compare the quality of VoIP from different providers? Or is sophisticated test equipment required for this?

This package seems to have good respect in the industry. Never used it myself, but you were asking for tools:

http://www.voipmonitor.org

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#2045054 27-Jun-2018 21:57

Foiler:

Customer is connected by Vodafone fibre 50/10 to Voyager's VoIP service .. just to clarify, the delay in one direction is about 0.5 sec on a mobile phone call.

You would think there might be tools available to compare the quality of VoIP from different providers? Or is sophisticated test equipment required for this? Something along the lines of an audio ping to a loop-back handset and then measure the round trip delay and distortion plus noise would be a good start.

You will always have a delay from a landline to a mobile in part because of the codecs used by mobiles. Jitter buffer settings in your phone will also affect this. I don't really understand what the issue is - are you getting packet loss or out of order packets?

VoIP doesn't create "distortion plus noise" because it's fully digital. As a result of this analysis of a VoIP call doesn't even look at audio because this is irrelevant. Analysis looks at the main key things - jitter, packet loss and latency, which together create MoS stats.

VoIP quality between providers won't vary in a way that can be measured in a way you think it can be.

Foiler

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#2053066 11-Jul-2018 07:35

sbiddle:

VoIP doesn't create "distortion plus noise" because it's fully digital.

Hmmm .. I have two customers with distortion and noise - regular loud squawks, background continuous noise. Not to mention gaps and delay and dropped calls and misbehaviour of the Cloud PBX (wrong phones sometimes ringing, can't transfer etc etc). Both businesses are with Conversant/Voyager and both had good service in previous years ie before the merger.

Similar experience here earlier this year? https://www.geekzone.co.nz/forums.asp?forumid=43&topicid=230794

jnimmo

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#2053070 11-Jul-2018 08:08

Sounds like transcoding issues to me..
are the phones locked down to using g711a?

Foiler

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#2053134 11-Jul-2018 09:28

jnimmo: Sounds like transcoding issues to me..
are the phones locked down to using g711a?

no they are using G729a

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#2054526 11-Jul-2018 19:42

Foiler:
sbiddle:

VoIP doesn't create "distortion plus noise" because it's fully digital.

Hmmm .. I have two customers with distortion and noise - regular loud squawks, background continuous noise. Not to mention gaps and delay and dropped calls and misbehaviour of the Cloud PBX (wrong phones sometimes ringing, can't transfer etc etc). Both businesses are with Conversant/Voyager and both had good service in previous years ie before the merger.

Similar experience here earlier this year? https://www.geekzone.co.nz/forums.asp?forumid=43&topicid=230794

Umm you have bigger issues than just distorted audio, looks like conversant / voyager have a bit of work to do, guess you also do not have a CIR on your connection also.

Edited, miss wording

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sbiddle

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#2054546 11-Jul-2018 19:57

Foiler:

jnimmo: Sounds like transcoding issues to me..
are the phones locked down to using g711a?

no they are using G729a

Ugh. Why would you be doing that?

jnimmo

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#2054773 12-Jul-2018 09:38

Switch to g711a - you should get perfect calls with that. It would be a good idea to switch to Voyager for UFB which will put them as close as possible to the voice switch.

This is standard in NZ to hand off calls between providers. Even 10 years ago we found g711a was far more reliable than 729 over an ADSL connection

Foiler

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#2054863 12-Jul-2018 10:43

The preferred Codec setting on the phones is G729a but an inbound call connects with G722 for both encode and decode, and an outbound call connects with G711a for both encode and decode.

Decode latency varies between 90 and 190 msec for the outbound/inbound calls respectively, but the round trip audio delay is measured roughly at 0.5 sec by one party saying stop to the other party counting one-and-two-and- ...

Unfortunately the Cisco SPA525G2 is reporting the R Factor, MOS-LQ and MOS-CQ fields as Not Available.

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#2054873 12-Jul-2018 10:59

sbiddle:

Foiler:

jnimmo: Sounds like transcoding issues to me..
are the phones locked down to using g711a?

no they are using G729a

Ugh. Why would you be doing that?

Sadly we still have providers that prefer G.729 over G.711a&u

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