OT: VOIP interfacing for recorded 'radio' show

[Dave Labrecque]

Right now, we're using the host and caller audio available at the headset jack (RJ-9) of the IP phone through some isolation transformers into and out of a small mixer. Actually it works really well, EXCEPT that one in five or so calls manifests a problem whereby the host comes out of the mixer sounding muddy and noisy, while the caller sounds fine. My guess is that there's something akin (in the POTS world) to a bad null that's letting host audio come back in on the caller signal. We're doing a proper mix-minus out of the mixer, BTW.

Anyway... is their a proper way to handle VOIP phone interfacing for 'radio' (this is actually an Internet radio/podcast talk show). Is there a special IP phone hybrid for such an application? I tried a standard POTS type hybrid and it didn't work.

Not trying to be a jerk, but anyone with actual knowledge preferred. ;) I've had enough of (me and other) old school guys guessing at how to solve this. That said, I guess I'm open to anything at this point. So ignore that last comment. :p

[CurtZHP]

Sounds like a bad null for sure. VOIP can be unpredictable in this area. I've found that, ever since telephony went completely digital, you're often at the mercy of whatever they send you, whether voice or data, when it comes to this stuff.

There are VOIP specific hybrid interfaces, but they're expensive. Not sure what your budget is. Telos Systems makes a few. I've never used them, but I have used their standard POTS units with great results. I suggest you call them and leave a message for Ted in tech support. He's been in the business for over 40 years and really knows his stuff.

[bcorkery]

Not trying to be a jerk, :pWhat a jerk! :p I can't add to what Curt said. I always ask for Ted at Telos. He's likely to shed some light on your problem even if you don't buy a box.

[Dave Labrecque]

Thanks, guys. Well, no one at Telos seems to know what to do with the name 'Ted'. Would you have a last name? Thanks!:)

[bcorkery]

Ted Alexander ... Hmm, any relation to Ian?
Anyway, you might have to call the help line (216) 622-0247.
This line may be specifically to those who have their equipment.
You can also contact them at: support@telosalliance.com.

G'Luck.

[Dave Labrecque]

Ted Alexander ... Hmm, any relation to Ian?
Anyway, you might have to call the help line (216) 622-0247.
This line may be specifically to those who have their equipment.
You can also contact them at: support@telosalliance.com.

G'Luck.

OMIGOD. I called the wrong Telos (shoulda just used the number you gave me) and spent a very unpleasant five minutes figuring that out with a woman who has no business representing any company as a phone answerer. She actually called me a 'dick'. Nutty. At least now I know why no one over there had heard of Ted.

I now have a call into Mr. Alexander (who, it turns out, has both a radio and voice-over history according to his bio on the site; maybe he's Ian's dad after all). Will let you know how it goes.

Thanks, again, gents.

[Sean McCoy]

She actually called me a 'dick'.
Well, at least she didn't call you a jerk.

[bcorkery]

She actually called me a 'dick'. Nutty.Wow1 Was it Telos Trucking Company????

Nobody representing any company should every say that to a customer or potential customer.

Yeah, Ted has chops. Wait until you hear his voice! He's pleasant and really knows his stuff.

[DominicPerry]

OMIGOD. I called the wrong Telos (shoulda just used the number you gave me) and spent a very unpleasant five minutes figuring that out with a woman who has no business representing any company as a phone answerer. She actually called me a 'dick'. Nutty. At least now I know why no one over there had heard of Ted.


I've had a bad day, and that's cheered me up. Something very amusing about the whole wading in with a request to the completely wrong company. And then being insulted. Thanks Dave. :):):D:)

Dominic

[Dave Labrecque]

Well, at least she didn't call you a jerk.

That woulda been okay. Especially if I had been one. ;)

[Dave Labrecque]

Wow1 Was it Telos Trucking Company????

Nobody representing any company should every say that to a customer or potential customer.

Yeah, Ted has chops. Wait until you hear his voice! He's pleasant and really knows his stuff.

Got a call back from Ted within an hour or so (of calling the right number ;)). As predicted, he was very friendly and way helpful. And they didn't actually have a product that could do what I needed (yet). But he jumped online and started googling stuff. Before we were done, he found something called a VOIP adapter, which is something he doesn't normally deal with. I thanked him for his outstanding, uncalled-for assistance. I think I have a path forward now. The idea would be to use a VOIP adapter to feed a POTS hybrid. More research to do, though. Truly a first-class cat, Ted Alexander is. Ian, you must be very proud. :D

That was after I'd written an email to the support address at the first Telos about the unprofessional manner of their operator. I got very specific with my complaint, outlining what she'd said to me. I got a call from them, too. The woman at the other end had read my email and agreed that the operator's behavior was completely inappropriate (I would hope so!) and apologized on behalf of the company, indicating that woman number one would be spoken to. Hopefully, the operator gal will get a-lickin'. And I don't mean the fun kind. :p :o :)

[Dave Labrecque]

I've had a bad day, and that's cheered me up. Something very amusing about the whole wading in with a request to the completely wrong company. And then being insulted. Thanks Dave. :):):D:)

Dominic

LOL. Happy to be of service, Dominic. :p

[CurtZHP]

We use a VOIP adapter at our station to interface our Telos TwoX12 hybrid to our digital phone system. Ours is built by Avaya, and it's pretty expensive because it has to convert a dozen phone lines. Basically, these things fool the hybrid into thinking it's see a standard copper pair POTS line. Works like a charm. I don't know if they make anything for just one line.

[Dave Labrecque]

We use a VOIP adapter at our station to interface our Telos TwoX12 hybrid to our digital phone system. Ours is built by Avaya, and it's pretty expensive because it has to convert a dozen phone lines. Basically, these things fool the hybrid into thinking it's see a standard copper pair POTS line. Works like a charm. I don't know if they make anything for just one line.

That's the ticket. Looks like something like this (http://www.voiplink.com/Grandstream_HandyTone_701_p/grandstream-701.htm) into a POTS hybrid might be the solution.

[RBIngraham]

Dave, thanks for the enjoyable thread. Remind me not to piss you off. :p

[Dave Labrecque]

We use a VOIP adapter at our station to interface our Telos TwoX12 hybrid to our digital phone system. Ours is built by Avaya, and it's pretty expensive because it has to convert a dozen phone lines. Basically, these things fool the hybrid into thinking it's see a standard copper pair POTS line. Works like a charm. I don't know if they make anything for just one line.

Curt, I know you're in no position to guarantee results, but would you say that I should be pretty darn confident that using a VOIP adaptor into a POTS hybrid should work fine? Anything I should check out before taking the plunge? Thanks.

[Dave Labrecque]

Dave, thanks for the enjoyable thread. Remind me not to piss you off. :p

Lest I call your spouse/partner/mother to complain. ;) (I assume your boss would not be particularly helpful in such a circumstance.)

[CurtZHP]

Curt, I know you're in no position to guarantee results, but would you say that I should be pretty darn confident that using a VOIP adaptor into a POTS hybrid should work fine? Anything I should check out before taking the plunge? Thanks.



I can tell you that it won't blow anything up if it doesn't work.

[Dave Labrecque]

I can tell you that it won't blow anything up if it doesn't work.

All righty then.

[jcgriggs]

Lest I call your spouse/partner/mother to complain. ;) (I assume your boss would not be particularly helpful in such a circumstance.)

Assuming you can find the right phone number :D

Cheerz,
John

[Dave Labrecque]

Assuming you can find the right phone number :D

Cheerz,
John

Hopefully, RBI would help me out with that. Flawed logic? :eek:

[Mattseymour]

Just a thought, but before you spend money on an analogue telephone adaptor try a softphone as you'll probably have a PC kicking about you could use. Assuming you've got a standard SIP server you're talking to, then try Jitsi (https://jitsi.org/) or you could also use an iphone/ipad I've had excellent results with Bria. There will be options for Android too. Much easier to configure than one of the Grandstream adaptors (which always give me a headache).

[RBIngraham]

Hopefully, RBI would help me out with that. Flawed logic? :eek:

LOL.... but my number is really not that hard to find. But I am afraid my wife probably wouldn't care and my mom has had a couple of strokes, so she would mostly just be çonfused about why you were caling her. :)

I do have bosses. Although as you probably suspect... I am not the person that answers the phone at any of the organizations I work for. I only answer complaints about the hearing impaired systems... :)

[Dave Labrecque]

Thanks to everyone's input on this. Looks like we're going to move ahead with a VOIP adapter (aka, an 'ATA') into a POTS-type hybrid. I've read that both the VOIP protocol and codec used will impact call quality and reliability (as well as latency, I'd guess).

Can anyone offer any pointers on which protocols/codecs would be best for this application? Or at least point me to a source of information about this? Having a little trouble tracking it down.

Thanks.

UPDATE: upon further reflection... I'm going to be pretty much stuck with whatever the VOIP provider is using for protocol and codec, I think. Is that right?

[Mattseymour]

The protocols will be sip for the signalling and rtp for the audio.

You're best going with g.711 (u or a, doesn't matter but most likely u if you're in Americaland). This should sound better than a pots line. You're right that it's really down to what the VoIP provider supports but g.711 is pretty universal. It's an uncompressed codec, 8-bit, 8khz sample rate giving you 300-3400hz in 64Kb/s. Avoid g.729 which is compressed low bit rate, 6-8Kb/s. It's ok for some uses but not for broadcast.

VoIP won't give you any degradation of quality, in fact the quality should be better, unless you're losing packets or the VoIP provider is trunking over g.729 to cheap out on bandwidth. If you experience poor audio quality then change the VoIP provider, robotic voices and stuttering, change your ISP or VoIP provider. If packet loss is a problem, and you can't fix it, see if GSM is supported. It doesn't sound as good as g.711 but GSM was designed for mobile phones so copes well with packet loss.

Latency also shouldn't be a problem. VoIP setups tend not to introduce any problematic latency beyond what's present in the network connection. So if you've got nice short ping times to the VoIP server all should be well. If your route to the server takes 500ms then conversation is going to be tricky. That said you can configure most VoIP devices to put more data into a packet, which lowers network overhead at the cost of greater latency, so don't do that. Some server implementations have an auto sizing jitter buffer that will increase latency if packets start arriving out of order.

We do all our office calls over VoIP now, most using public internet. I can do a loopback test over 3G to our server and the latency is low enough to be inconsequential in conversation. Bit weird when talking to yourself though.

[Dave Labrecque]

The protocols will be sip for the signalling and rtp for the audio.

You're best going with g.711 (u or a, doesn't matter but most likely u if you're in Americaland). This should sound better than a pots line. You're right that it's really down to what the VoIP provider supports but g.711 is pretty universal. It's an uncompressed codec, 8-bit, 8khz sample rate giving you 300-3400hz in 64Kb/s. Avoid g.729 which is compressed low bit rate, 6-8Kb/s. It's ok for some uses but not for broadcast.

VoIP won't give you any degradation of quality, in fact the quality should be better, unless you're losing packets or the VoIP provider is trunking over g.729 to cheap out on bandwidth. If you experience poor audio quality then change the VoIP provider, robotic voices and stuttering, change your ISP or VoIP provider. If packet loss is a problem, and you can't fix it, see if GSM is supported. It doesn't sound as good as g.711 but GSM was designed for mobile phones so copes well with packet loss.

Latency also shouldn't be a problem. VoIP setups tend not to introduce any problematic latency beyond what's present in the network connection. So if you've got nice short ping times to the VoIP server all should be well. If your route to the server takes 500ms then conversation is going to be tricky. That said you can configure most VoIP devices to put more data into a packet, which lowers network overhead at the cost of greater latency, so don't do that. Some server implementations have an auto sizing jitter buffer that will increase latency if packets start arriving out of order.

We do all our office calls over VoIP now, most using public internet. I can do a loopback test over 3G to our server and the latency is low enough to be inconsequential in conversation. Bit weird when talking to yourself though.

So, you can choose your own codec without regard for whatever codec may be used on "the other end"? I guess that makes sense, cuz it's all just packets of data in between. I'm used to ISDN, where both ends have to be the same codec. Now I wonder why that is.

Matt -- thanks so much for the info. Very much appreciated. Let's see how we do... :)

[jcgriggs]

Dave,

VOIP uses SIP (Session Initiation Protocol) and SDP (Session Description Protocol) to negotiate the settings for a given session.

Basically, as part of setting up a call, one end of the call sends SDP packets (inside an SIP stream) that describe all of the codecs and settings it supports (ordered by preference) and the other end sends back an SDP reply where it identifies the settings that it supports, selected from the provided list and generally favouring the initiator's preferences. So, while you do have some freedom to select codecs, etc., you still have to match options that are supported by your service provider and the other end point for the call(s).

Hope this helps.

Regards,
John

[Dave Labrecque]

Dave,

VOIP uses SIP (Session Initiation Protocol) and SDP (Session Description Protocol) to negotiate the settings for a given session.

Basically, as part of setting up a call, one end of the call sends SDP packets (inside an SIP stream) that describe all of the codecs and settings it supports (ordered by preference) and the other end sends back an SDP reply where it identifies the settings that it supports, selected from the provided list and generally favouring the initiator's preferences. So, while you do have some freedom to select codecs, etc., you still have to match options that are supported by your service provider and the other end point for the call(s).

Hope this helps.

Regards,
John
Interesting. So, I may/probably will get "what I want", but there may be occasions where it goes to the second or third on my list in order of preference?

So do I understand right that to improve chances of using the codec I want, I should definitely initiate the call? That would be fine in this scenario, since she does guest phone interviews and typically calls the guest interviewee just before show time.

[jcgriggs]

Dave,

Here are some links that may help you understand how SDP and codecs are related.

This one describes how SDP is used when setting up a VOIP call:

http://www.3cx.com/blog/voip-howto/sdp-voip2/

It is actually the second part of an article. The first part gives details about what an SDP packet contains:

http://www.3cx.com/blog/voip-howto/sdp-voip/

Cheerz,
John

[Mattseymour]

As John says, both ends have to support the codec and this is all worked out in the sip session. Your voip provider will be able to tell you what's supported. Your device's preferences might be respected but often the server preferences win out so if you want to use a particular supported codec it's sometimes worth turning all others off.

[Dave Labrecque]

As John says, both ends have to support the codec and this is all worked out in the sip session. Your voip provider will be able to tell you what's supported. Your device's preferences might be respected but often the server preferences win out so if you want to use a particular supported codec it's sometimes worth turning all others off.

Good stuff. Thanks, guys. :)