| Recordings using WinPcap Hi, when I record a call on my server using WinPcap; the way I sound when I play back, is that exactly what the recipient hears? |
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| WinPcap just captures packets; I'll assume that you are using Wireshark to create a .au file.
If you are capturing at the point where the listener's IP phone is located, if there are no network impairments, they should sound about the same, except for obvious differences caused by the transducers, e.g. most computer speakers have much better bass response than a typical phone. However, on a network with bad jitter, the recording will sound unrealistically good, because packets that arrived too late to be "aired" and were discarded by the phone, will still be present in the recording. In case of packet loss, the opposite can be true; Wireshark simply replaces lost packets with silence, while some phones have sophisticated "error concealment" algorithms.
If you are capturing at the source of a stream, the audio that arrives at the destination can of course be degraded by the network in various ways. In that case, a good recording merely proves that the trouble was not caused by the source device or server. |
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| reply to pacpac
Understand, I thought for a moment I could use WinPcap to also measure call quality on the receiving end....... My friends are getting a bit tired of my test callls  |
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| You can analyze a packet capture with the Stream Analysis features of Wireshark, or export the capture into a spreadsheet, script, etc. for more detailed analysis.
Depending on the nature of the trouble, leaving yourself a long voicemail at the remote end, then retrieving it by non-VoIP means, is a good way to evaluate outbound voice quality.
For some types of trouble, starting a test call while a problematic one is in progress, will give you useful data without annoying your contact.
In some cases, it's useful to call back into your own system via a SIPBroker / iNum / Localphone / Rebtel or (ugh) Betamax gateway in the destination country. Unfortunately, this usually is not useful for mobiles, often routed by a different carrier. |
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| reply to pacpac
Thanks, good ideas. When I call the echo test, *777, to my 3CX system from my location, does the packet travel from my ATA to my server and then back to my ATA? If so, would not this give a fair indication of both call quality and latency? |
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| The echo test should accurately show latency. However, using it for quality testing on a ATA can run afoul of the ATA's echo suppressor, which attenuates outbound audio during strong inbound audio. The suppressor is needed, because limitations imposed by the codec make it impossible to cancel all the echo (that a remote party would otherwise hear). If you have an IP phone, use that instead. Otherwise, speak in short bursts, i.e. you go silent before the echo starts, and don't speak again until the previous echo is over. |
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| reply to pacpac
Very useful indeed. So, I understand that when I do *777 the packet goes directly to my server and back (not via my VoIP provider). When I try this I get about zero audible latency. I could then reasonably conclude that the VoIP provider POP should be as close to my server as possible? |
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| If your VoIP provider does not proxy the media, the location of their POP doesn't matter, because your voice packets will go directly to/from the provider's upstream carrier or DID carrier.
I know almost nothing about 3CX, but it may have a setting that re-invites the audio, so it flows directly from your ATA to the VoIP provider or his upstream carrier. Using such a feature requires careful configuration of NAT mapping on the ATA, etc., to avoid problems with no or one-way audio. From your "zero audible latency" description, it appears that you won't need this. |
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| reply to pacpac
Thank you very much for providing valuable input...
After reading a series of posts on this excellent forum, done a bit of reading/research, and live testing I have come to the conclusion that using a VoIP provider who do not proxy media is the optimal for my set-up. In summary:
a) Running 3CX on a dedicated server with Static IPs in the UK. 3CX gives me all the IVRs, voice mail, transfers, etc. as I require.
b) ATA connected to 3CX is located in the US, South America and the UK. Calls are primarily placed in and to these locations.
c) Set up own domain with DNS records enabling Direct SIP incoming to 3CX from DIDs.
Configure 3 VoIP providers in 3CX, the first 2 do not proxy media and the 3rd do proxy media.
Now I need to select the best quality, no-frills and reasonably priced VoIP provider for outgoing calls. Need no other features than just calling out. |
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