I'm researching IPOffice for my company and it was unclear to me whether the voice traffic is transmitted using IP or not. Or is it basically a PBX with IP capabilities?
The reason I ask is I'm not really thrilled with VOIP quality at this point and I'd like to go with a traditional digital system, so you're saying IPoffice can be configured to not use IP at all?
I'm glad to see other folks are researching this solution as well so I don't feel all alone My company has a Merlin Legend at the HQ and a Merlin Magix at the satellite location. The sites are conencted using tandem networking. The Legend has an Adtran box breaking out voice and data, while the Magix has an INA board doing the job. The connecting circuit is a full DS1.
If we add another satellite location I am looking to upgrade the whole shooting match to IP Office. Drop an IP 406 at HQ and an IP 403 at each satellite location.
What I am curious to know if any folks in this forum have had good experiences implementing IPO in similar situations. I am looking for exclusive VoIP up until HQ hands off external calls to the PSTN through a PRI circuit. Each satellite would commumicate with HQ using a point to point DS1. Of course I would have a small loop start trunk pool for backup communication at these sites.
Past that I was thinking that having IP softphones as an option would be nice for laptop users as well as PDA users. That and the Outlook/Exchange voice mail interface seem to be nice luxuries.
But back to my original question. Have any of y'all had experience installing IPO using exclusive VoIP for such tandem networking? Just curious since the product line is relatively new for Avaya. I know back in 2001 when I was provisioning my INA board there weren't many folks out there in the same boat. I had to pull old Lucent PortMaster docs off the Internet and fly a bit blindly...
IP office networking through a dedicated data link & Voip works well with good feature transparency.
I would not recommend using IP phones as standard around a site as thier Admin & overheads can be akward, dedicated digital handsets are a much better option.
However for roaming & remote users IP Softphone is an Ideal solution, & remote user with a dialup networking connection can be "in the office" even when at home
Thanks for the feedback. I was wondering about terms of VoIP overhead. If I was have exclusive IP phones I know that it would add to LAN traffic. Our average LAN utilization is only about 1-3% over a given length of time. Of course that can burst higher depending on what's going on, but it has a low baseline.
If we were to have about 40 IP phones at one site you still think this wouldn't be a good idea? I thought that the VCM resources only come into play when IP-to-IP calls are being setup. IP to PSTN calls occupy one VCM channel for their duration. If I am looking to implement a 30 channel VCM at the HQ location this should be reasonable, right? If not then I will reassess things and perhaps look to have physical desksets be digital phones as you recommended.
Within the LAN/Wan environment the IP Phones comply with H323v2 standards. Which means they make use of the standard codec’s (Voice Compressions Techniques) which are built into H323v2, The standard for Voice and Video over IP.
There are 2 main Codec’s that you would consider when using IP Phones.
G.729a and G.711
G.729a is normally used across WAN/VPN connections because of the restrictions in available bandwidth. G.711 is used in the LAN environment, because you normally have at least 100MB to each device in the LAN environment, so bandwidth is not so much of an issue. G.711 does not compress the voice stream just encapsulates the 64k PCM stream in to IP packets.
A standard call using G.729a across an Ethernet network, with no additional compression will consume approx 31.2 kbps
A standard call using G.711 across an Ethernet network, with no additional compression will consume approx 87.2 kbps
There are no problems with installing 40 IP Phones at one site however you need to work of how many concurrent calls you want to be able to allow between the IPO and the IP Phones and provision the VCM card correctly.
Your statement is correct
I thought that the VCM resources only come into play when IP-to-IP calls are being setup, once set up it drops VCM channel because the Phones support there own compression. However a call from an IP Phone to a digital / trunk / pstn will consume a channel for the duration to the call
You should also consider getting some decent Ethernet Switching devices that support COS @ Layer 2 and @ L3 to make sure VoIP traffic get priority across the Ethernet network. The AVAYA Cajun switches are designed for this. You would be looking at creating a VLAN'ed network.
Also make sure the IP Phones that you chose support all the features that you require, you will normally not get as many features on an IP Phone compared to a 20XX or 6400 digital phones. The product description has a good matrix of all features supported across all phones. Think of an IP Phone as a fancy alog phone with some additional features.
IP Phones in a correctly configured lan enviroment are as MrIPO Says not a problem - yourealy only need as many VCM channels as there are external lines to prevent blocking.
My recommendation against 100% IP phones is mainly from convinience, we have 1 site like this & have had the following minior problems
1) the DHCP server in the IPO is crap & will issue duplicate addresses if multiple requests are recieved at the same time (eg 5 or more IP Phones trying to reset after a system reboot)
2)when resetting the IP phone trys to download fresh firmware, this requires a tftp server (Manager app) & can take some time to complete a reset.
3) any problems on the data network could result in a total loss of telephone conectivity
dedicated digital telephones do not suffer from 1 & 2, & a mix of standard & IP phones would reduce the impact of 3
In my experience with IP quality, even uncompressed voice (64k) still sounds sibilant. You hear it with S's and Z's and other consonants that make similar sounds. It's just not as clear as a digital PBX. Why should we be taking a step back in quality in the office environment? Maybe the cell phone has dropped people's threshold for acceptable voice quality. Right now IP is a great solution for keeping remote users in the system, but it can't subsitute for digital in the office quite yet. IP is a really good minor leaguer that got thrown into the major leagues a season or two too early. I'd like to hear from those that disagree with me...
I have a question regarding IP Agent for which I am hoping to get help. I am new to IP Agent and Office. One of our overseas sites has IP Agent (they do not have a PBX there, they are linked with ours. Currently all calls originate in our Definity Multivantage and are routed to the overseas site via IP Agent) and I am wondering if they can continue using IP Agent if we connect our two sites via DS1 trunks, i.e., stop using IP connectivity to route the calls. Is it also possible to connect analog phones to the PCs running IP Agent?
Cgull - what IP Telephony platforms have you heard ugly G.711 calls from? I have been implementing IPTel networks for over 3 years now (thousands of IP phones) and have not had this experience with G.711 codecs (ever.) Now, G.729a and G.723.1 are a different story, but even G.729a has become acceptable while admittedly lesser in quality than G.711 calls. One scenario where I have observed what you are describing was not the fault of the voice codec in use, but gain (actually negative gain, or attenuation) in the voice gateways not being set properly (too hot, not attenuated, and with a near perfect signal....OVERDRIVE.) Haven't ever observed anything but toll quality, digital phone quality when phone to phone over a LAN or even WAN with the proper bandwidth.
Talbotpat- I wouldn't call the G.711 calls ugly, but like I said it wasn't digital quality. This was a Cisco-based client site of a vendor that was used as a real-world demo for us. A few years from now we may not be having this conversation, but I am typically late to jump on tech bandwagons, especially when my first hand experience is below my standards. The 2 IP-only vendors I've talked with have subtly or agressively tried to "scare" me into their solution because "this is all there will be in 5 years", but when I balked one of them told me he'd pitch me 2 other IP-enabled PBXes as well. It's not gigabit compatible, which granted isn't the deciding factor, but it throws out their push for a single cable to the desk. Regardless, I have to buy a phone system now, not a few years from now. I have to base that decision on the demos I've seen and heard and the client referenes of these vendors. One of the great things about calling references for phone systems is you get to somewhat test out the system while you call. I found the quality of the other end ranged from acceptable to horrible, but even the best sounded a little crunchy.
G.711 has always treated me great. When using P2P T-1 circuits to connect sites, 711 is above toll quality IMO. I have even been surprised of the quality level of 729a, for VPN sites with limited bandwith you can't ask for better quality with its small draw.
Agreed. The testing scenario you describe brings much more than the voice codec into the mix. Most of which have no more to do with an IP handset on a IP telephony solution than they do a digital handset on a TDM solution.
A standard call using G.729a across an Ethernet network, with no additional compression will consume approx 31.2 kbps
A standard call using G.711 across an Ethernet network, with no additional compression will consume approx 87.2 kbps
I'm told that G.729a uses 8kbps. Which is correct and why(anyone else feel free to toss in their two cents)? This makes a big difference to me as we are on DSL and having some issues, and I suspect bandwidth problems between sites.
G.729 produces a voice packet of 20bytes 50 times a second which is as you correctly say 8Kbps.
But for those voice packets to get anywhere you have to add an IP Header (another 20bytes), a UDP header (another 8 bytes) and an RTP header (another 12 bytes).
So now your G.729 8Kbps has become 24Kbps.
And then the network type (Ethernet LAN, Frame Relay, ...) add a bit more. The ethernet header is another 14 bytes, and so we've reached the 30Kbps level that MrIPO stated. The 8Kbps gives you just compressed voice data but the rest is necessary if you want to actually get that voice anywhere.
Does it get worse, yes, if you want this to be a two way call allow for another 30Kbps coming back.
This is nothing to do with the IP Office either, its the same for any VoIP traffic across Ethernet.
The good news is that most Ethernet pipes (eg. from the phone to the switch) are pretty fat (10/100Mbps). Which is good as the IP Office doesn't do QoS on the LAN side.
And when it goes onto a WAN link, even though they are much narrower, the WAN headers are much more efficient than the Ethernet headers (on a WAN PPP link with header compression the original 8Kbps voice only becomes 10Kbps). Plus the IP Office does do QoS on the WAN side.
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