Wireless Networks Thread, Getting Network ready for VOIP in Technical; We're going to be getting implementing a new VOIP Phone System soon, and I am now starting to prepare the ...
13th December 2011, 11:38 AM #1
Getting Network ready for VOIP
We're going to be getting implementing a new VOIP Phone System soon, and I am now starting to prepare the network for this. All our switches are HP Procurve, with a 5406zl at the core with the edge switches made up of 2510, 2610 & 2626 (we're about to add various new PoE switches in certain locations as well). I'm putting in place a VLAN specifically for the VOIP (and tagged this as Voice) but now scratching my head with regards to QoS. I've yet to talk to the engineers of the kit we're going for (Splicecom) to find out exactly what is expected, but would appreciate any insight in doing this.
I've been going off looking at various bits and bobs on the web, with some settings specific to certain systems but any help would be gratefully recieved. I fully understand that I need to prioritise the VLAN with the VOIP traffic, but would welcome any other persons experiences and settings they implemented so I don't have any issues when it starts going into place.
Thanks in advance.
13th December 2011, 12:07 PM #2
There's a lot of horror stories out there on VoIP.
We started with ours on a flat LAN with managed switches, but no VLAN, setting the phones to prioritise their RTSP traffic using DSCP set to "EF". You may need to tell your switches to honour this, but I doubt it. We never had a moments bother out of it.
We currently route all internal calls over IP, including those to our charlotte NC office. This is over a VPN, on some links which are very much non-traffic controlled. It works fine.
I imagine we might run into trouble if one of the choke points became very congested.
I have a sneaky feeling many VoIP horror stories are from vendors wanting to sell expensive kit
Thanks to tom_newton from:
FragglePete (13th December 2011)
13th December 2011, 12:23 PM #3
Thanks @tom_newton. Yes, I've read varous horror stories too hence why I ensuring that I've got everything right in the first place. The backbone here is pretty good, installed 2 years ago during a refurb so I'm fortunate to have a true star topology of all the switches.
As a temporary measure, I installed an Elastix Box with about 5 Grandstream phones just so we could get phones to areas that needed them desperately and have been impressed with the quality of the calls, and these just sit on the network as is.
It's a certain amount of nerves I guess, this is a big thing for the School and it's got to be right and it's sitting on my network!
13th December 2011, 01:03 PM #4
A lot of the horror stories about VoiP telephony come from bad implementations.
If you keep everything VLan'd sort out your switch tagging properly and get your DHCP sorted you should have no problems especially if your using an IP enabled PBX gateway to the PSTN
However if your also moving carrier to an IP enabled SIP service then its a different story.
Your better off to have a dedicated DSL service with a voice grade carrier for your SIP Trunks as trying to share your existing Internet connection can be fraught with problems.
We have Mitel IP Phones everywhere in line with the PC's network jack dedicated VLAN HP Switchgear trunked kinks between each switch with QOS that gets us a bullet proof connection to the IP PBX from there we have ISDN over Fibre for our link into the TalkTalk network.
Each site has it's own Fibre gateway and IP PBX however they are linked together over the MPLS to allow the internal routing so internal extention to extention calls stay on our LAN/MPLS rather than going public. If either ISDN route goes out each PBX can route out through each other.
Now thats the really good VoiP system.
The other is a small 10 user call center using Grandstream/Yealink IP phones they have to use a shared Internet connection to a SIP provider (Soho66)
Over all, for what they pay the service is excellent but it can and does occassionally drop all calls and the phone will loose the SIP server for a few minutes and occassionally due to Internet traffic the voice quality can vary a lot.
I get 2 or 3 complaints a week regarding the SIP service but they have got used to this and learned to live with it as it saves them £2k per month in call charges!
So it really all comes down to what you need to use it for and how you implement it but the same rules apply, Keep it simple.
Thanks to m25man from:
FragglePete (13th December 2011)
13th December 2011, 01:26 PM #5
Thanks for your insight @m25man. We're not using SIP trunks, this is going to be connected to our existing ISDN30 line that School has. I'm currently setting up the VLANs on each of the Switches in preperation, and tagging the existing uplinks (trunks) between the Core and Edge switches. Documenting as I go! Some locations are going to need additional switches (which are on order) and I'm going to make use of the spare fibres we have on some of the runs to add dedicated uplinks (sorry, trunks! ) where the number of phones is quite high.
It's more the QoS side I'm interested in at the moment; a lot of reading up on implementations of VOIP and ProCurve seem to point to setting DSCP to 101110 (46) on the Voice VLAN which seems easy enough to implement but just wanting to know what others have set.
Again, many thanks
13th December 2011, 01:37 PM #6
Find myself nodding at post by @m25man! We do a similar thing, our 3 Zultys systems (Leeds, Charlotte, Soton) all connect to the wider world via PSTN - fractional ISDN30, plus some regular phone lines as backup/overflow. All internal calls are over VPN, where the VPN connection is generally a dedicated ADSL except here in Leeds where everything goes over our 100Meg microwave link from metronet. Yes, VoIP... over wireless. Its fine.
Bear in mind G711 is ~64kbit/sec/call. It's nowt for today's decent LAN, especially if your switches are making sure the little packets move quickly!
Thanks to tom_newton from:
FragglePete (13th December 2011)
13th December 2011, 01:51 PM #7
We have a new school being built at the moment and the area has no copper to the building only fibre, so the option of PSTN is out and they have to go down the SIP route ... and I am reluctant to have voice and data on the same line ... partly because the data could end up going with a non-edu ISP!
13th December 2011, 01:54 PM #8
I've only just started VLANing our network, and have had VoIP on it a while with no issues of call quality due to network, no QoS enabled etc. We also have IAX2 trunks to an external provider who handle our PSTN connection. Until last week we've had no issues with this (out over 100Mb to POP, the 1Gb to DCs then multiple 1Gb to wider networks then to external provider's network), have just started seeing some issues on inbound call quality (lost or dropped packets) which I'm going through the process of trying to work out where the issue lies. But overall, have seen no issues with not much work done!
13th December 2011, 01:57 PM #9
We have got lines coming in via a fibre connection for one of our Norstar systems provided by Kingston Communications.
15th December 2011, 01:02 PM #10
Dont quite understand all these horror stories, one of the places we support has Cisco VOIP installed without any issues on HP ProCurve network.
Switches we use are ProCurve 2910al-48G-POE+ and some 2610-24-PWR - VOIP has its own vlan:
vlan xx qos priority 6
If you do get any 2910al's make sure the latest firmware is updated on them!
Thanks to IanT from:
FragglePete (15th December 2011)
15th December 2011, 05:34 PM #11
Trying to work out the QoS stuff at the moment, looking at the Splicecom stuff it states to following:
Which has really got me scratching my head, I've seen lots of references to DSCP 46 which I can set with a DSCP Policy of 101110, but not sure about 40. Unless I've got myself all muddled up. Any help would be appreciated!
• Setting the SERVICE TYPE of every voice IP datagram to the value of 0xA0 (10100000)
• When using the DIFFERENTIATED SERVICES interpretation of the SERVICE TYPE, the CODEPOINT (DSCP) is set to 40
• When using the older TYPE OF SERVICE (TOS) interpretation, the PRECEDENCE is 5, D is 0, T is 0, and R is 0
10th January 2012, 10:09 AM #12
I've used Splicecom in the past, excellent kit.
How are you installing it, are you going for a central mdoule with expansion modules in cabinets that allow you to use your existing analogue phones on the cat 5 data lines, or are you replacing all handsets with IP handsets ?
10th January 2012, 10:49 AM #13
Just one Call Server with pure IP Handsets around the school (120 in total!). Currently running around the School setting up all the switches and identifying ports that will be on the 'Voice' VLAN. Just installed the Premium Edge License on the Core Switch to enable Multi-Casting across the VLANs (PIM Dense).
Originally Posted by budgester
Engineers on-site from Monday next week. Quite excited. My only query at the moment is the issue where we have rooms that don't have enough ports in. Here the PCs will be plugging into the back of the Phones but not sure how to set the config on the ports for this. I'm assuming that the port will be 'Untagged' on the main Data VLAN, 'tagged' with the Voice VLAN and the phones will tag the voice traffic.
10th January 2012, 10:58 AM #14
Just Seen this post - You can run ISDN 30 over Fibre - We currently have ISDN30 DASS over fibre!
Originally Posted by GrumbleDook
With regards to the VLANing. I've setup the Voice Vlans on all our Switches in preperation but havn't quite got the system in yet - We are going with Hosted VOIP (with as yet undecided carrier) over a leased Line.
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