Audio IP Codecs (Part 2) Or, How Telos ZIP One is NOT the new ISDN
Well, much has changed since I last wrote about my quest to venture into the IP Audio market. You can read all about it in part one (Audio IP Codecs Part 1). So a few things: It turns out the ISDN market is not exactly dead as it were in the Northeast as originally thought. Dave Immer up at Digifone wrote all about the Verizon ISDN redeployment on his blog, which you can read here: (Dave Immer’s Verizon Blog Post)
Other important developments
During that last writing, I reached out to most of the IP Audio Codec manufacturers (both Hardware and Software-based) for connectivity/compatibility assessments. My connectivity tests were based on the v1.7 software (the first available update from Telos for the ZIP One) to feature RTP connections with the Luci Suite of IP Software. Sadly, it was also the LAST working version from Telos (they are now up to v1.9.42b) that allowed for the very important SIP Configuration. This all important feature allowed for the registering of SIP Server credentials where you could effectively get your SIP connection “bridged” or middle-hosted thereby traversing the so-called Firewall Port Blocks. It was amazing. It allowed me to generate and test connections with many codecs. Alas, the subsequent upgrades to the Telos ZIP One rendered this feature all but dead. It’s a shame really, as most of my communications with Telos have yielded nill on this front. So, in short I can not honestly recommend the Telos ZIP One for Voice Talent looking to embrace technology with the widest versatility and interoperability.
Browser-based IP Codecs
My work with the Browser-based Audio Codecs were initially rejected out of hand because of I wasn’t sure I could achieve the same or better quality as with ISDN. But that has changed in a BIG way. The second reason was: I could not (for the life of me) get AUDIO ROUTES via my DAW. Sounds crazy I know. But the idea of me sitting at a computer in front of a Mic tracking my own audio and patching in Directors/Producers did not sit well with me. So, Sound-Streak, ipDTL, and Source-Connect NOW (more on these two below) seemed proprietary and with no interoperability with other tech did not present itself as viable options either.
I recently worked with two great and extremely helpful guys Rob Marshall (of Source-Elements) and Kevin Leach (of ipDTL). Both were kind enough get me up to speed on the two add-on technologies they offer:
Source-Connect NOW offered by Source-Elements takes advantage of a plug-in for ProTools (or any DAW I would imagine) the effectively route audio via the Chrome Browser. An awesome inclusion, as I can basically send out a Guest invite for NOW and connect via ProTools to either provide the audio tacking on my end, and/or the Clients end.
ipDTL offered as another ISDN replacement has effectively OPENED its platform with the advent of SIP connectivity. In fact, as I write this plans are being made to test the service from my Telos ZIP One. Currently there is a How-To guide that has been developed for the COMREX ACCESS IP Codec. But, again as with “SIP Compliant” the idea is that if its SIP-compliant, it “should just work.” Unfortunately, if the ipDTL requires that I use the SIP Configuration on the Telos ZIP One, I’m shit outta luck – as that feature no longer works 🙁
So, my efforts had me reach the following Companies/People with mixed results. The Email excerpts are below:
First up was TIELINE:
From Tieline: (note: Mary Ann Seidler has since moved on from Tieline back to Telos, her former home)
“Hello Mr. Lang,
Thank you for downloading our audio experts guide for IP. I hope you find it a useful reference.
We are just starting to work with a couple of voice over artists who are using IP codecs to replace their ISDN codecs.
If you would ever be interested in being a part of that demo project, please let me know. We would be honoured to have your input on the technology.
Mary Ann Seidler
Vice President – Sales Americas
Mary was instrumental in pairing me up with one of her tech guys at Tieline, and even offered to put one a Tieline Codec my desk to test) to get a Test Line connection established. It went off brilliantly! I was able to establish a SIP send/receive connection with the test line without a hitch. Remember, the process was to look for the lowest common denominator for Voice Talent. What would be the easiest, most compatible, most versatile to run Audio IP sessions for voiceover work?
Next Up: Audio-TX/STL-IP
I can’t say enough good things about this Company and Mo Dutta. In addition to being extremely helpful and well versed in the technology, these guys have kick-ass products to boot!. I reached out to Mo and his team many, many times. When I was originally in the market for my ISDN box, I contacted Mo. When it came time to look into the IP market I sadly (and am kicking myself now) overlooked this company – thinking they were more software focused and I wasn’t ready to run Windows-based software on my Mac yet.
Turns out they were far, far from a one trick pony: Their new software implementation of the IP Audio software is indeed SIP Compliant, has over 10K users, dual ISDN/IP for a connection directory, and they are constantly keeping ahead of the curve.
Here was my communications with Mo:
From: Nathan Lang
Sent: 30 July 2013 14:35
To: AudioTX STL-IP
Subject: A u d i o T X S T L – I P Questions…
Hello, I’m a voiceover talent in the US. I’am putting together tech information for the VO community with regard to IP codecs, and I thought I’d reach out to various IP Codec manufacturers with a few questions regarding interoperability.
Might someone take a look at the questions and provide answers to the following:
1) Is the AudioTX STL-IP SIP compliant, and if so are there test lines for testing out non-Audio TX IP Codecs?
2) If not, then what sort of connection algorithm is used with the AudioTX STL-IP in terms of auto-negotiation
3) Does this Codec connect to other non-AudioTX equipment?
Thanks so much in advance!
And Mo Dutta’s follow-up:
Hi Nathan –
Head of Sales, MDOUK
Tel. +44 (0)1886 883900
Fax. +44 (0)1886 883909
Sure enough as in fact was the case, connecting with the AudioTX-STL was a matter of my doing a minor configuration on my end, and Voila! I was up and running with a UK Producer I usually work with via ISDN or Source-Connect.
Next Up COMREX:
I posed the same question(s) as above (sent to all the manufacturers I reached out to) and here was the response.
Thank you for your message regarding your Comrex equipment.
1 Our ACCESS and BRIC-Link products are SIP compliant. Please refer to the following application which covers a few SIP solutions that we do recommend.
2 Generally G.722 or G.711 are used for SIP connectivity with our equipment. Depending on the application itself, our units can work with other algorithms.
3 We can connect, via SIP, to other equipment, however, we can not maintain a guarantee of that since we have no control of their level of compatibility. If you would like further detail, please let us know and we will try to provide you some additional support on the matter.
Paul @ Comrex
19 Pine Road
Devens, MA 01434
Here again, I was able to not connect at not just G.722 and G.711 (minimum reserved VOIP connectivity) – but MPEG L2 128K as well!. Now, as paul said there is never a guarantee that the connection will be maintained for any length of time. But it is telling that you can in fact dial in your setting and at least see if the receiving end will at least attempt to “match” the callers end. So at a minimum interoperability was achieved.
Next up was MAYAH Communications – The IP Codec Company:
Well known for their superior ISDN/APT Codecs, there was NO question in my mind that their game wouldn’t be at the highest level. They did not disappoint. Below was my query along with their response from Armin Woods:
Please see my answers below your questions in red.
Consultant international Sales
Mobile +49 151 22933650
Hello Armin, I’m a voiceover talent here in the US. I’m putting together tech information for the VO community with regard to IP codecs, and I thought I’d reach out to various IP Codec manufacturers with a few questions regarding interoperability.
Might you take a look at the questions and provide answers to the following:
1) Which Mayah IP Codecs are SIP compliant, and if so are there test lines for testing out non-Mayah IP Codecs?
All of our codecs are SIP compliant (make sure they are on latest version and firmware)
2) If not, then what sort of connection algorithm is used with the Mayah IP in terms of auto-negotiation
You can also go direct IP if the dialed devices feature a public IP address
3) Does this Codec connect to other non-Mayah IP equipment?
Yes it does
Thank you Armin. Can you send me the SIP address test lines for your company?
Next up was Technica Del Arte – Makers of the fantastic Luci Live Software
Hello, I’m a Voice Actor in the US, and I’m currently putting together tech information for the VO community with regard to Audio IP codecs, and I thought I’d reach out to various IP Codec manufacturers with a few questions regarding interoperability. I bought the Telos Z/IP One Codec a few months ago with the intention of using it as a backup/alternate for ISDN sessions. Now I find the doesn’t easily connect and “play nicely” with other Audio IP Codecs. I’m aware that my Telos Z/IP v 1.7 software is supposed to work with a Luci product, and I have become somewhat familiar with “SIP” connections and so what I’m writing to ask is basically:
1) I believe the Luci products to SIP compliant. Would I be able to achieve the same “broadcast-quality” connections that I currently enjoy using my ISDN Codec, and if so how?
2) What are the differences between the different Luci products.
3) Are there test lines for connecting to any of the Luci Audio IP products?
4) If not, then what sort of connection algorithm is used with the Luci Audio IP in terms of auto-negotiation.
5) If in the end, I needed to make a recommendation to Studios, Producers, other VO Talent, what which product would that be, considering it would be Studio-To-Studio connections (as is my case) and not necessarily connections from say, a call from a phone in the field back to the Studio?
Any other information you might provide that hasn’t been addressed here would be much appreciated.
Thanks so much in advance!
I got a response back from Joost Bloemen at Luci Live:
HI Nathan, below my answers:
1) Yes, we have high quality AAC-HE and MP2 codecs, you can use these with SIP also.
2) Luci Live is for use in the field. Luci Studio is for in the studio, receives connections via RTP only, which works also at places where SIP doesn’t work.
3) Yes, we have an echo server that lets you connect and hear yourself back.
4) We can use RTP or SIP. The both auto-negotiate. many ip audio codecs like Luci Studio now automatically recocnize incoming RTP-streams.
5) Studio to Studio, i’d use Luci Studio, it has many many high quality codecs, liek high bitrate AAC,AAC-HE, MP2, Linear, Flac , ULCC, even 24-bit is possible.
Hope that helps, please keep me in the loop.
I never did get the information for the Test/Echo server test connection, but I can confirm I have setup Luci Live Lite on my android phone and dialed back into my Studio and it works flawlessly!
And last, but certainly not least, I contacted Doug Irwin Chief Engineer for Major Market Radio Networks after reading his article:
Doug wrote an article on radiomagazineonline.com regarding SIP Codecs and I reached out to him as an expert on the subject:
Hello Doug. My name is Nathan Lang and I’m a voice over actor in NYC. I recently came across an the article you wrote for Radiomagonline, and I found it interesting that while you mentioned the various manufacturers for for IP Codecs, I found no mention of “interoperability” among the devices. I bought the Telos Z/IP One earlier this year looking to integrate it into my studio as a backup/alternative for the ISDN connections I now use for point-to-point live sessions.
After a good bit of time (and having reached out to Telos, Tieline, Comrex, Mayah, AudioTX and Musicam CCS), I found ways to manage SIP connections – but I’m still trying to find the equivalent connection rate and quality I currently achieve via ISDN.
I wrote this piece back in July chronicling my efforts:
So, my question is… Where might a non-technical person continue to look to find integration solutions for Codec interoperability? I’m familiar with the EBU/NAICP specs for compatibility, but I find the IP Audio Codec world very much like the beginning of the ISDN world insofar as the devices not being able to “play together.” The impression seems to be that the manufacturers want you to “lock in” to using two of the same devices for connections as opposed to a IP devices being able to just work with one another.
What are your thoughts?
HI Nathan, below my answers (Doug Irwin’s Response)
First, thanks for reading the article and yes, you’re correct; I didn’t say too much about interoperability this time around. I did some research and wrote an article about the very same topic 3 years ago, in case you’re interested:
http://radiomagonline.com/studio_audio/codecs/radio_nacip_simplifying_codec/ but it sounds to me like you already know quite a bit about it. (Actually you fit my definition of ‘technically-minded’ person rather easily.) Figure 1 in that article shows the codecs that are supposed to work (between different manufacturers) when they claim they’re N/ACIP compliant. (I just read your piece and I see your concerns about Telos.)
I can’t say I know of anyone that is doing what you are doing, though, so in fact I can’t really offer up any suggestions. The people I knew back in NYC (I was there till about 6 months ago) were using Zephyrs for home studio applications of course. In the year 2013 it’s a little hard to believe but I think you’re plan on using IP for audio transport for VO is, in practice, a little ahead of its time.
I wish I had a better answer for you Nathan. Actually, if you make some progress, I’d like to hear about it.
Thanks again for reading Radio.
Earlier today during a Google search I found a post of SIP, and thought hmmmm, this seems like an interesting topic to chime in on. It was on the well-respected and all around good guy and voice talent – Dave Courvoisier’s Blog:
Hello Dave, I wrote up an article (http://iam.nathanlang.com/ipcodecs) on my blog back in July of 2013 somewhat lamenting the demise of ISDN service in my area. NYC was hit hard by superstorm Sandy and Verizon had problems at its main copper site. Being no stranger to adventure, I purchased the Telos Z/IP One and sought to work out other connection options outside of ISDN my other go-to tech: Source-Connect. I set out to find an alternative specifically for me and others like me: Voiceover Actors. As you might expect, the foray was not without headaches!
In any event, I learned a great deal about SIP and possibilities that the future holds for it. Turns out that much like the ISDN interoperability standards, this tech was (sort of) mandated to guarantee minimum connections with other hardware/software SIP codecs. Widespread use of this technology (while not there yet) could ante up the user base considerably. I reached out to all the major tech manufacturers (both software and hardware alike) to establish minimum working protocols for SIP and RTP connections. Let me tell you, the results were fantastic.
Still, if Studios, Engineers, Directors and Producers don’t embrace the technology sadly it will be relegated to its original intended audiences: STL – Radio Station Studio-Transmitter-Links. Another part of the problem is this: Manufacturers want to insure that you stick with THEIR codecs. So, if you buy a Telos ZIP one, Telos can guarantee rock-solid connections with other ZIP ones. Thankfully, Kirk Harnack and the other guys at Telos keep their ears to the pavement ensuring a well-rounded interoperability plan. I think until ALL the tech plays together nicely. ipDTL connecting to say, a hardware codec (which I’m told there are possibilities now). Until my codec can dial up a SIP or RTP address on say, Skype or Source-Connect NOW, or dial into Mo Dutta’s STL-IP codec. Or, Sound-Streak allowing for SIP connections – well then, this all stays proprietary. Not very useful when its goes up against tried and true ISDN.
Thankfully, bridging services exist like Dave Immer up at Digifon that can link these disparate services together. But truthfully, Producers just want to book in this day and age without any additional drama.
All of the manufacturers I reached out to offered (if it were possible) various ways to establish minimum VO connections (MPEG L3 48K) I use via ISDN. The ones that were realistic were clunky at best, needing a SIP server to establish the connection. Then, there are the firewall (NAT) issues. Lets just say we will leave that for another post. So, in a not so short post – YES this technology is real, YES this technology works. Once the interoperability standards are fully worked out, and actors don’t have to buy into proprietary hardware/software it will be a much different conversation. I urge all vo actors not yet familiar with this tech to reach out with any questions.
In short, I think this SIP/IP/RTP thing is the real deal. Just a matter of more standards in terms of interoperability. I don’t mean G.711 or G.722 either. But real “Open Standards”. None of this proprietary “I will only play with this device” nonsense. The real way forward for the manufacturers and other industry players if getting everyone on the same page. Much like my Prima LT being able to connect with my Producers Zephyr then connect to the other Talent in DC’s Mayah. Small achievements like that, you know?
Feel free to email me to test for IP compatibility with my Telos ZIP One codec.
UPDATE February 3, 2015:
I was given SIP credentials from Kevin Leach at ipDTL the open platform now makes it easy to connect via a SIP connection. Sweet! Kevin was kind enough to get me setup to make the connection much the same way that the COMREX ACCESS device can – using SIP Registration Credentials. I talked about this previously. This is where you pretty much enter the registration information that you do for VOIP calling, or IM’ing via Apps on your computer. Think Skype, Adium, Yate, etc.
Well lo and behold. This credentials CANNOT be used on the current v2.0 version of software on the Telos ZIP One codec!
It was suggested I go back to v1.7 (with the bugs and all) to use the SIP Configuration feature, which was the ONLY version where it worked. Nice Telos! So, the current version LOST the ability to connect with virtually ANYONE, ANYWHERE via SIP, but they did NOT forget to include the $600 Add-on License for the APT Feature, which I understand comes standard on some of the Hardware Audio IP Codecs. Here again is why I’d say to anyone optioning to go the Hardware route – to take a look at Comrex, Tieline or Mayah. Musicam is no longer done here in the states (NJ) but still a good choice. For a software option: Audio-STL IP.