So this is my attempt to gather some real data here to support anecdotal eveidence that suggest POTS, Skype, whatsapp is better than comercial VOIP, SIP, RTP, u-law ... etc. I would love to find a real solution to get POTS quality phone service in my home but short of stringing a wire between two cans (which has very little latency by the way), I have given up on that.
To measure the final REAL mouth to ear delay across a phone call I built a few elecret mic amps to drive a two channel osciliscope like so:
Which resulted in test traces with the two mics near each other and me mouthing a tick sound like:
Ok. Lets test. I have two Grandstream GXP1405 VOIP phones with jitter buffer set to 100ms (minimum setting) and "adaptive". These are both on the same LAN and relay through Asterisk. I did tcpdump packet captures in and out of Asterisk and found the delay to be less than 5ms for the PCM u-law packets going in and out.
Thats over 100ms added by, I assume, all the fancy codec s/w+h/w. Clearly a string and two cans (or POTS) beats all this digital technology hands down. I will try a few other VOIP phones to see if I can get a unit with zero jitter buffer.
Now a test with the echo service at my VOIP provider voip.ms. Before doing this I did do packet captures looking at the delay between VOIP phone and ISP CPE. The path was: GXP1405 VOIP phone outgoing packet --> Asterisk --> Router --> ATT DSL modem incomming packet. Total delay contriibuted here was <5ms. The round trip ping time to the local voip.ms pop is 13ms and no other traffic is flowing through the ATT DSL modem. I bought this connection specifically for VOIP due to the very low jitter characteristics (less than 2ms jitter looking at smokeping) it has over a cable connection. Hoping this would fix my VOIP latency problems - but alas....
Thats 160ms ROUND TRIP on the echo test. Given the LAN-2-LAN test this leads one to believe the "problem" is mostly with my VOIP phones and not the VOIP provider. I would love to hear this answer and a recomendation on what VOIP phones to buy. I have old Polycom and Cortelco phones. Oddly the oldest and cheapest of the later seem to give the best (anecdotal - have not tested yet) quality call.
Next test is a real world test. A call from VOIP phone to a cell phone. In this case the path is GXP1405 VOIP phone --> Asterisk --> Router --> ATT DSL modem --> 'Net --> Voip.ms --> ATT Cell.
Wow!! >300ms ONE WAY (from VOIP phone mic to cell phone speaker). This is noticable and annoying to the callers. I am not sure where the delays are here.
As a final test I decided to test the latency of a WhatsAPP call as anecdotal evidence has told me that thes calls are much better than any SIP/RTP call I ever make. So using the same mic amps and osciliscope setup I tried whatsapp between two iphones connecting over LTE between ATT and T-Mobile carriers in California.
Thats 250ms one way. A little less but comprable to the "bad latency experience" call from VOIP/SIP/RTP and cell. Why? The sound quality of either call is good (no static, clicks, or cyber-sounding codec problems) but am still consistently told the latency/interaction quality of the non-standard-SIP calls are much better - even if the call is from Russia to US. So i guess more study is needed here. But one thing is for sure - give me less latency given a good 'Net connection. Otherwise I guess my colleagues and I need to go get POTS lines or switch completely to proprietary VOIP apps like whatsapp/skype.