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Hosted VoIP (Asterisk) call quality issues

Posted on 2007-11-20
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Last Modified: 2013-12-27
Hi,

I am experimentally rolling out hosted VoIP in around ten small offices in the UK and Europe and one or two in North and Central America, but I am dogged by call quality issues in some locations whereas in other offices it's almost faultless. Some calls simply cut off, which I gather is most likely to be a NAT Proxy related issue, and some calls are really distorted and broken, which I gather is most likely a network connection/packet loss/latency issue. Strangely the remoteness of the geographic location does not always correlate with poor service.

As we need to make use of the Asterisk PBX, all the traffic is directed through the hosted VoIP servers in London as opposed to going point to point once connected. Obviously this adds to the call quality problems in certain locations. Also most of the handsets are behind firewalls using NAT so a NAT proxy is required which, as i understand it, further adds to the problems.

I know that the simple answer should be to install a Asterisk box on each site, but I'd like to avoid this if at all possible especially as some offices only have 2 or 3 extensions, and I have not found a cheap enough piece of hardware which will manage the site and that can have an IAX trunk to our provider.

I have been using Linksys SPA941 & SPA942 handsets and am really happy with them. Routers vary office to office and the internet connection is mostly ADSL & SDSL, but as I say there are realtively few extensions and internet demand from other devices in the offices is minimal.

ALL tips and pointers are extremely welcome!


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Question by:gabiosz
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by:bhnmi
ID: 20324479
I mean transport can be an issue also. The is no traffic shaping if you are traversing a public network (most likely multiple networks) to reach the PBX.
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by:gabiosz
ID: 20324491
Can you elaborate on that for me?
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bhnmi earned 750 total points
ID: 20324530
If you have site a and b both with different ISP's connecting back to the host pbx on a different isp then a and b the is no QoS between the sites and this means your traffic is treated like everyone else's. The ideal way to connect multiple locations using VoIP servers located in geographically different places is MPLS.
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Author Comment

by:gabiosz
ID: 20324595
How do hosted VoIP companies get around call quality issues such as this, surely there is a way of doing this over the public network, or it pushes the cost up dramatically?

I appologise, I have become lazy over the past few years and my networking knowlage has suffered, I haven't really had to concern myself with Internet traffic in any detail for quite sometime.
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by:grblades
grblades earned 750 total points
ID: 20326390
VoIP requires a very good internet connection with very low packet loss. Just 2% packet loss will be very noticeable while on a VoIP call but almost undetectable for general internet traffic.

We have an Asterisk server in the office and a dedicated internet connection for it. We have sales people who travel around the world and connect back to it to make their calls as it is significantly less expensive than hotel or mobile rates. We also use a VoIP provider so anyone can choose to make a call via them.
Most destinations worked fine however there were a few that always caused problems (china for example)

I am on NTL internet (UK) at home and I could not use the voip provider due to the NTL router the traffic went through was dropping the occasional packet. However I could connect to the office system without any issues.

There is software called MTR (WINMTR on windows) which performs a traceroute and pings every router in the path. You could use that and ping the hosted server and heave it running for about 1000 pings and that may give you an indication of where the problem is occuring. Feel free to post the results here if you would like me to interperate the results.

If you are having problems due to the NAT devices in the remote offices then the symptom will be that either people cannot dial you (unlikely) or that the call gets established but the remote office cannot hear any audio)
I am a bit confused about which protocol you are using. The Linksys phones use the SIP protocol but you mentioned about using an IAX trunk.
If you are using IAX to a device in the remote office and then have the phones connect to the device via SIP then you should not have any issues. IAX works much better through NAT and only requires port 4569 be forwarded.
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Author Comment

by:gabiosz
ID: 20328559
Thanks for your input.

What I was saying about IAX was that as a last resort we'd like to use a simple box on the LAN to deal with the internals there, one which is capable of using IAX tunk to connect to our VoIP suppier. I have looked at the Linksys SPA9000 but think it looks a bit restricted and I'm not sure if it runs on asterisk but the price is right. Again bear in mind we are looking at between 2 and 6 phones per office, so cost effectiveness is key.
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Expert Comment

by:grblades
ID: 20328688
The SPA9000 does run Asterisk internally but you cant get access to the internals. Everything is configured through their own web interface so the usability is more limited. I am not sure if it supports IAX.
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Expert Comment

by:bhnmi
ID: 20328844
I could help myself. I am waiting for my Switchvox (Astrix Based VoIP Platform) right now and this post make me want to play with it even more. ;P
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