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HiPath 4000 V4 IP voice quality problem

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badrikali

Technical User
Mar 12, 2009
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Hi,

We have a HiPath 4000 V4 with STMI4 module used for connecting openstage 40/60/80 phones.

The problem that we face is the voice - when an IP extension user calls another person, if the 2 parties talk simultaneously, the IP phone user cannot hear anything that is said. It is readily reproducible, when we call the local announcements on the PTT like time or weather and talk. When we talk, we dot hear the announcement. We have the same problem even if we have just the telephone, the system completely isolated from the IP network. We do not have this problem with optipoint 500 phones.

The local Siemens tech says we have the latest software for the system and the STMI4 board and the openstage phones.

Has anybody any else got the same problem? Any solutions?

Regards
Sundar



 
Check and make sure the ext`s have DMC activated in SDAT.
 
Sorry discard the last reply i didnt read through your post with much attention.....appologises for that ;)

does the fault occur if both party`s are ip devices?

 
You can't compare the IP behaviour of your Openstage's with your TDM OptiPoint 500's. IP is not TDM. What you are hearing (or not hearing) is a side effect of VOIP, it's not a fault.
 
Hi,

When we did the same testing with HiPath 4000 V2 the problem is not so bad. We still can hear the other person, but in version V4 - it is seriously bad!!

Regards
Sundar
 
I've had the same behaviour on UV2.0. UV4 will be a different echo cancellation engine on the phones/stmi's I guess, but nevertheless, it's not really an error. Whether you can persuade Siemens to accept it as a fault depends on how good you are at complaining, but that problem won't be eliminated. If you really can't live with it, you'll need to put a TDM phone in.
 
This is a truly ignorant question but I'm throwing it out there anyway....

Would this be an issue related to duplex settings? Like on a cheap conference phone (or older Rolm phones) if one person is speaking the other cannot interrupt - because the phone only supports half duplex, but if you go to a more expensive or more fully featured phone it supports full duplex communication and end of problem??

I'm thinking along the lines of some kind of duplex setting on the gateway or some other attempt to limit bandwidth that's causing the phone to exhibit this behavior...

Sorry for the relative dumbness of the question, but everyone needs to learn someplace - and I might have one of these beasts installed next year...
 
It's a good point. You need to check it's FD on both sides of the lan connection. If you set auto on one side and not the other, you'll negotiate a HD connection. You need auto both sides, or FD fixed both sides. It's definitely worth checking. If the phone is set 100MBFD, it doesn't guarantee a FD connection, you need to check the data switch too and see what it's set the port to. It might be that simple....
 
Half Duplex settings affect the bandwidth available by effectively halving it.
Having a link on Half Duplex has no discernable effect on the voice quality except a very very slight increase in average latency (you'd have to be looking pretty hard to spot it) due to the occasional delay caused by contention.

For example, you have a phone on a 10Mb half duplex link - this means that you only get 5Mb each way. As you phone only needs (say) 80Kb, this has no effect as your 10MbHD interface is perfectly capable of passing traffic at 80Kb in both directions.
The thing that causes highly noticeable voice quality issues is a duplex mismatch.


Half Duplex does *not* mean only one side can speak at once! People who tell you this are confusing two entirely seperate layers of the OSI model.
 
Hi,

In a test we did we just had the STMI, HP4K, and 2 openstage phones and the problem was very much felt.

Tried to set the room properties to "Echo" was very marginally better. But, the problem still remains.

Regards
Sundar
 
Did you check the duplex settings on the STMI, Openstage and the switch ports, or not ? If you did, and they are the same, then it's a characteristic of the echo cancellation required for VoIP and you're going to have to live with it.

I was wondering if you'd upgraded from UV2.0 to UV4.0 in which case the tool won't necessarily convert your old STMI config to the new CGWB, and thus it's possible your NIC settings are different, it's worth checking.

Why do you want to speak at the same time anyway ? That's just rude.
 
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