atcomsystems.ca/forum Forums TDM Business Telephone Systems Toshiba Telephones & Systems CTX and BIPU

One of my clients has a CTX670 that his phone vendor has put BIPUs in and now says it is doing Qsig across the network? The problem is that calls are being 'dropped' frequently. This vendor is blaming the network, yet calls are being dropped at the District Office as well that don't even touch the network.

As the network vendor, I am trying to set QoS but neither the client nor the vendor can tell me how the Toshiba is marking packets. From my investigations, it seems that this system is not really doing VoIP - he uses the same phones he always has and my QoS doesn't pick up anything even when tuned to the IPs of the phone systems.

Can anyone explain any of this to me? What marking is the PBX using to set ToS on the packets? My fair expereince with VoIP is that the when the network is causing problems it results in stutter, 'in-a-barrel' and that kind of thing given connectivity?

Any insight into this BIPU and how it is working woudl be a big help. This phone vendor has lost every contract in this area due to sloppy work and refuses to up any info.

Thank you in advance.

Toshiba uses either 802.11p or DSCP for voice tagging depending on how the programmer set it up. Normally we don't enable it because if there's anything between the systems that doesn't support QoS, you won't get any voice.

Qsig is a protocol for linking 2 or more PBX's together. You can use the same TDM phones for intercom and outside calls, the only time VoIP comes into play is calls between PBX's.

Dropped calls can be due to high network traffic not allowing enough bandwidth to sustain the voice call. Are you using point to point links or VPN over the internet?

Point to point and heavily used. I'm working now to QoS the PBX traffic, minimize the existing traffic on the links and some other queuing and stuff. I can affect all traffic except the PBX.

The confusing part is that the central location drops the most calls and the traffic to and from that location wouldn't touch the network.

Monitoring indicates that 3 out 4 of the locations not having problems are in the top 4 most congested links. Other than this (contradiction) there is no correlation between link usage and problem frequency.

If it IS disabled, then is it correct that the network won't be able to affect the voice packets? Have you had any success manipulating this traffic at layer 3 (IP)?

You've been fantastic help already. Thank you.

You say that calls are being dropped at the central location. If you have a hub and spoke setup wherein the hub is the 'central location' and the remote locations are using the trunks from the central location for inbound and outbound calls, I would start looking for the cause at the central location. What type of trunks do you have? What do you experience when the call is dropped? e.g. fast busy, clicking, etc. What does the display read when the call is dropped? Does it go immediately back to displaying the date and time?

What are you using for calls coming into the central location, PRI? If so you can have PRI slipage which is affecting one of my customers now. Some calls start a bit of hum which gets more pronounced and then the call drops.

When you are figuring bandwidth, you have Xk transmit, Xk receive and "overhead." If you had 23 channels at 32K you would need more than 23x32K because of "overhead." I allow at least an extra 10% bandwidth and more if possible to avoid that problem.

You can use an Adtran or Kentrox QoS router to take incoming bandwidth and guarantee adequate bandwidth to the voice section of bandwidth usage.

We are using a VOIP provider who gives us a PRI output from their server. Two options:

Data T-1 to server input, PRI output to Toshiba RPTU
Covad SDSL to server input, PRI output to Toshiba RPTU

If data bandwidth has to be split off:
Data T-1 to Adtran, QoS output to server input, PRI output to Toshiba RPTU and non-QoS output to data router and switch.

Sorry, got pulled to something else. My thing is the network and I do most of that remotely. So here's their answers to my questions.

Does it mean that they pick up the phone and there is no dialtone?
Yes

or are they talking and the call just hangs up? Yes

Or do they dial and then nothing happens?
Yes. Sometimes the display says: "Call Abandoned"

Does it always happen on outbound calls or inbound? Both

Does it happen both?
Yes

Is it always the same at each site or does the problem present differently at each site? Each site is the same. They all exhibit the behavior of all these symptoms.

Each call?
No. Random. Sometimes 5 in a row, sometimes hours between them. 8:00 - 9:00 seems to be the worst. Massive phone/data volume at that time.

Also, does anyone have an answer to the DO call questions *below?
Yes. The same as the other sites. Inbound and Outbound, 4-digit and
DID
*The DO?! Are these always calls out to the other sites? Or just normal calls?


Lines are all T1s muxed out of a T3 with an Adtran 830 (I replaced the old 800, to no avail).
Qwest has 'tested' the lines, at least at the DO. I've suggested they ask that both ends be tested, they hadn't thought of that!
I'm told the trunks (4) are PRIs and see that in the Adtran.
Most of the other lines are data T1s to the sites, which work well. On a handful of these they are still muxing voice on a couple of channels and oddly enough these sites are low-to-no on the drops! The rest of the sites are connecting using the BIPU--Q1A-R(the 'Q1' is hand-written), which I understand to be the Qsig version? Even stranger, one of the 4 cards is marked with 'M2' in the hand-written part? But there are no IP phones, so I am told.

On the network side, I've garaunteed 500K for traffic coming from the IPs addys of the BIPUs. The routers at the far end don't have that ability, so it's from DO to remotes. They highest I've caught so far is 300K.
I haven't heard if it made any difference but I'm not expecting it.

Everyone's giving great info. Thanks a million.

Tekdif, your woes are just too complicated to try and fix on this forum. I would suggest you have a Toshiba technician come on site and get with tech support. There are trace logs that can be ran to help determine the cause or causes of your problems.

Agreed, phonemeister! Mysteriously the aforementioned vendor has committed go onsite and stay until it's fixed. Funny thing is they are taking a new proecessor and some other hardware. I think my poking with a sharp stick has jarred some sort of memory or realization and they are moving to cover now.

You guys are great! Thanks for everything. Always a pleasure to hash over things with some real pros.

If I was your Toshiba vendor I would certainly do some type of VOIP testing to check QOS. The problem with this is that most VOIP testing software is expensive or at least used to be. IXIA has Chariot testing which emulates VOIP traffic between multiple sites. This has come in handy for us when we are emplimenting VOIP on an exiting LAN or WAN. We have been able to put providers in check when they say it isnt the network or the router is configured correctly. There are so many reasons for something not working correctly and only a handfull of fixes. This software has gotten me and my co workers out of sticky situations and really slammed AT&T's hand in the door so to speak....