voip.ms discussion

[RamKat]

In preparation for my eventual switch to teksavvy I have started to configure an old WRT54G router to (hopefully)jive with my PAP2T and VOIP.MS.

I flashed the router with the VOIP version of DD-WRT. From comments that I have seen it may not be necessary to use the VOIP version (I would actually prefer the VPN version since that would provide me with another toy ....and learning curve)

Any experience out there or pointers on how to set DD-WRT up for VOIP.ms? QoS? SIP? Protocol? Priority? Ports?

[notsure]

RamKat, up until today I had been using dd-wrt for many years but I only started using voip.ms last March or so.

Getting the QoS just right with dd-wrt was a bit of a pain but it worked for about the last 4-5 months. Earlier this week we started having issues with choppy calls and I determined the problem was with the dd-wrt QoS. So, this morning I flashed my router over to tomato to give it a whirl.

The flashing took about 3 minutes and the basic configuring about the same. I had basic QoS working in another 5 minutes. Once that was done the graphing provided by tomato allowed me to tweek things (maybe 15 minutes playing around) and we now have great sounding call quality again even when my DSL line is under full load from other users in the house (5M/800k).

Long story short, if you want to learn I'd give tomato a try first. The price is right but your mileage may vary.

Cheers.

[RamKat]

Thank you Notsure. I came to the same conclusion but have not flashed over to Tomato yet (downloaded it to my PC though). I will probably do that next.

[canadavenyc]

I've been running voip.ms now on my PAP2T for a couple months now, using the Montreal2 server. I've run into a couple of weird issues that I'm wondering if anyone else has had?

First, I was on the phone to a buddy of mine in the USA who I hadn't spoken to in years, and we were talking for about 2 hours or so. All of a sudden I found that I was talking to dead air. Then I heard IVR phone "on hold music" (must've come from my end, because he uses regular landline service). Then it went dead. I hadn't run out of funds or anything, and the phone settings on voip.ms were not set to limit calls to a particular length (I just checked, it hung up at the 2hr 37min mark of the call).

That was a few weeks ago. A few nights ago, my grandmother called me from NYC. We had only spoken for a minute or so, and then suddenly the line went dead. A few seconds later, I got a voicemail email attachment, as if the line had suddenly been switched from "talk" mode to "leave a voicemail" mode.

I opened a ticket, and voip.ms got back to me saying to switch to the Chicago server, because "it's a newer server with newer software." I've just tried that, so we'll see how that goes.

But has anyone else experienced those kind of hangup or in-call re-route issues?

[pmosher]

I have two phones connected to the same PAP2T, registered to separate sub-accounts with voip.ms, receiving calls on different DIDs. Line 1 is specified as SIP port 5060, and Line 1 is on port 5061. Although everything seems to work, I'm puzzled about it because of the information in the voip.ms wiki that states ports 5060, 5080 and 42872 are the ones to use on their servers.

I asked voip.ms support about it in a ticket and this is the reply:

Please note that the port 5060 and 5061 that you're using with the lines in your PAP2T are only for internal use and to avoid conflicts between the lines, the port that is used to communicate with our server is the 5060. In this case you shouldn't have any issue with our service with your current settings.

I admit I don't get this -- it makes it sound like I could choose any ports I want within the Line 1 and Line 2 settings, and that somehow port 5060 would be used to communicate to voip.ms. Yet when I look at the status page on the portal, it shows Line 1 registered on 5060 and Line 2 registered on 5061, on the same server, the way I have configured it.

Can anyone explain what's going on, and why my configuration works even though it's using a port (5061) that isn't supposed to be open on voip.ms servers? I should say that the PAP2T is behind a Netgear WNDR3700 router. Is there some kind of NAT magic going on here?

Thanks very much for any insights!

Pauline

[Mango]

You've interpreted their (admittedly slightly cryptic) reply correctly. In your example, your device sends SIP traffic to VoIP.ms only on port 5060. VoIP.ms sends SIP traffic to you on ports 5060 for line one and 5061 for line 2. Your SIP ports can indeed be anything you want. VoIP.ms's SIP ports can be anything they want, and they have ostensibly decided on 5060, 5080, and 42872.

If you want to change the port your device sends SIP traffic to VoIP.ms on, you can do that in the Proxy field. For example: seattle.voip.ms:42872

I agree with them that your device appears to be configured properly.

m.

[pmosher]

Thanks for that very clear explanation, Mango. I don't need to become a network guru, but I also don't want to be totally confused about what's going on!

Pauline

[Gimli]

To put the process in even more laymen terms each line of your PAP2T contacts the voip.ms server using port 5060 and, during the registration process, tells the voip.ms server to contact them on ports 5060 and 5061, respectively, when calls come in for either lines.

That's also why you can't receive calls if your PAP2T isn't registered, the voip.ms server doesn't know where to send the call and on what port.

[canadavenyc]

OK, I just had ANOTHER abrupt hangup (actually my wife had the hangup) while using voip.ms. The phone went dead, she called her party back, and the person said that the line just seemed to go dead.

This is the third or fourth abrupt disconnect using voip.ms in the last month or two. I was thisclose to switching to voip.ms as my sole phone provider, since I really love a lot about it, but I don't think I can do it if there's going to be these random abrupt hangups.

I'm pretty certain it's not my ISP by the way--I have a reliable cable internet provider.

So, again, I appeal to the group--does anyone have any ideas as to what might be happening here? Everything works fine about 95% of the time, but the 5% of the time it hangs up on me kills me.

[marunio]

Canadavenyc,
I have similar experiences to yours and I've been with VOIP.MS as my sole provider for almost 2 years. The features, options and possibilities are great, but that 5% of hangups while on the phone, strange occasional crosstalk, or other issues are annoying. A question one has to ask himself is: am I OK with that 5% unreliability ?
m

[Mango]

It looks like you've already checked the obvious causes of this problem. If you don't mind some interrogation, let's see if we can get closer to the solution.

Is there any pattern to the issue? Does it happen on incoming or outgoing calls? If outgoing, value or premium routing? Does it happen with multiple phones? Does it happen with calls to/from a particular city/province? Can you make the problem happen if you disturb the cables connected to your ATA, or the cables connected to your phone?

If you've satisfied all these and you want to double check that your internet connection isn't causing your problem, download the free version of Ping Plotter and set it up to ping your VoIP.ms PoP a few times each minute. This will tell you if you have any brief periods of packet loss that you may not notice with regular internet traffic.

I see you've already switched PoPs and you're still having the issue, so internet routing is unlikely to be the cause of the problem.

Can you check your CDR around the time you were speaking to your grandmother, and verify that she didn't call back while you had your phone off the hook?

Do you hear the music every time the call drops?

m.

[canadavenyc]

Hi Mango,

Thanks...yep, I don't think it's anything obvious. And no, I don't mind interrogation

Any pattern? Nope, not as far as I can see. Not time-of-day related, nor length-of-call, nor incoming vs. outgoing, nor related to/from a particular call destination. Just completely random.

I can try to jiggle the phone/ATA cables a bit, but I doubt that a bum cable is the answer. I'll definitely check it out though, you never know.

I have a Mac, so can't run PingPlotter. However I'll have a look around, I'm sure there's a Mac-compatible equivalent somewhere.

I just checked the CDR, and when my grandmother called back, I did have the phone on the hook, not off.

I just heard the on-hold music once...the first time, when I was on the phone with my friend in the USA and it dropped a 2 hr 37 minute call.

I doubt these drops are caused by packet loss. It seems like too much of a coincidence for two calls (my grandmother's call to me, and the call a couple days ago outbound) to drop within a couple minutes of talking. I had a packet loss problem a few months back that was traced to a definite resolution by my cable company, and I did ping tests religiously for a week....my connection has been extremely stable. It's not impossible, of course, but I'd characterize it as unlikely.

[DanTou]

It has been happening to me too in the last couple of weeks. But it is a little different. While talking, one of the two parties can still hear the other sometimes. Before going dead, there is sometimes a bizarre noise. Most of the time, it just hangs up.

So, you are not alone.

If I used my phone for business, it would be unacceptable. As a home consumer, I can live with it. I call back, and I just tell people: "what can you expect for a dollar a month phone with all options?"... And then, they ask me how to get voip.ms.

It would be bad it it happened during an emergency call. But what are the chances? In my case, I have a cell phone fro emergencies anyway.

[Mango]

If one of the two parties can still hear the other, (which one?) then the call did not drop. That is known as one-way audio and is typically a router issue. What router are you using, DanTou?

[Jake]

Anyone know if it is possible to have the call logs automatically emailed or archived? It seems there is a 3 month limit.