- This topic has 24 replies, 1 voice, and was last updated 19 years, 3 months ago by Jitendra singh.
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7th January 2002 at 02:29 #11930Call Center ManagerGuest
Which is a good buy for PABX system? apart from Avaya? Anyone have comments on Ericson or any suggestion?
7th January 2002 at 11:58 #11931SANJAY BANGROOGuestIS IT FOR A CALL CENTER PLATFORM OR FOR YOUR OFFICE COMMUNICATION ??
8th January 2002 at 06:46 #11932Call Center ManagerGuestIt is for call center platform, Sanjay.
9th January 2002 at 05:46 #11933SANJAY BANGROOGuestThe best choices ,in order, are AVAYA , ALCATEL and NORTEL.
If u require any more clarifications do revert back to me.Sanjay Bangroo
13th February 2002 at 20:07 #11934Tim HarveyGuestTo save yourself allot of money and headache, you may want to look at the new VoIP (Voice Over IP) options. It will not only save you money on the equipment, but the compression rates for an IP call versus a standard voice call saves you about 5x’s the bandwidth for your IPLC requirements. The quality is almost identical to that of a traditional voice call.
I have implemented these solutions and would be happy to discuss this with you.Tim Harvey
14th February 2002 at 05:58 #11935SANJAY BANGROOGuestHi All ,
Find below a brief comparison of various voice technologies.
PLEASE REMEMBER FOR A PUR VOICE NETWORK TDM IS THE BEST CHOICE IN TERMS OF QOS, BANDWIDTH EFFICIENCY ETC. USING VARIOUS COMPRESSION ALGORITHMS .
G729A BEING THE BEST CHOICE FOR A PURE VOICE NETWORK.Cheers !!!!!
SanjayNETWORK DELAY AND JITTER
Key ingredient for good voice is
low delay
Delay can be measured as
Average length of time for packet to moveVariability in the arrival of packets
A jitter buffer nullifies the effect of variation
Latency is sum of above two
A latency of 200 – 250 msec makes a bad voice
Example consider network with 50 ms avvg delay but some part goes as high as 200-250 msec
A Jitter buffer of 10 msec will have low delay but will miss delayed packets
A jitter buffer of 200 msec will compensate for late packets but increase latecy
Hence Dynamic Jitter Buffering is a must criterion for Bandwidth ManagersCOMPARING TECHNOLOGIES
TDM
Quality of Voice : HIGHEST
Provides predictable and consitent quality of voice
Does not suffer from effects of Latency, JitterBandwidth Efficiency :
Very High for a pure voice network
Poor for a multiservice network
Cost : Cheapest
Effictive support for Multiservices: PoorFRAME RELAY
Quality of Voice :HIGH
Provides very good quality of voice under optimal conditions
Does not lower efficiency on implementing solutions for Latency, JitterBandwidth Efficiency :
High for a pure voice network
Very efficient for a voice/ data network
Cost :Medium
Effictive support for Multiservices: GoodVOIP
Quality of Voice : Optimal to poor
Provides very good quality of voice under optimal conditions
lowest efficiency on implementing solutions for Latency, JitterBandwidth Efficiency :
Very Low for a pure voice network
Not very efficient for a voice/ data network
Most efficient for data Networks
Cost : Medium
Effictive support for Multiservices: LowTECHNOLOGY FOR PURE VOICE NETWORK
Dynamic Bandwidth Allocation is not an issue
Voice Quality is of paramount importance
Bandwidth efficiency is significantly importantTDM is the best choice to meet the above requirement
TECHNOLOGY FOR VOICE /DATA NETWORK
Dynamic Bandwidth Allocation is an important issue
Voice Quality is of significant importance
Bandwidth efficiency is paramount importance important
FRAME RELAY is the best choice to meet the aboveDELAY OR LATENCY
Caused By:
algorithmic processing delay
frame delay (size of frame)
transmission delay
General Rule:
Delay approaching 32 ms requires echo cancellation
Delay is a function of transport technology alsoChoice of Technology has bearing on
1.Bandwidth Efficiency
2.Delay
3.JitterAND HENCE ON VOICE QUALITY
Choice of Technology is also governed by
Type of traffic running viz voice only or multiservices
Whether Dynamic Bandwidth allocation desired
QOS Requirement in multiservicesSOLUTIONS FOR JITTER
Variation in delay for Packet voice due to Network CongestionDoes not apply to Circuit Switched TDM
Happens when a large packet has already transmitted resulting in a voice packet waiting
Solved using techniques like fragmentation etc.BUT
EFFECTS OF FRAGMENTATION ON EFFICIENCY
Fragmentation results in creation of large no. of data packets
Each packet has to have a packet overhead
Packet overhead of FR is much lesser than IPIP overhead reduces WAN Efficiency by 10 – 15 %
FR overhead reduces by 2-4 %COMPARING PACKET TECHNOLOGIES ON QOS :
FR offers higher levels of prioritizationATM offers highest levels
IP is very limited in this aspect
TECHNOLOGY FOR PURE VOICE NETWORK
Dynamic Bandwidth Allocation is not an issue
Voice Quality is of paramount importance
Bandwidth efficiency is significantly importantTDM is the best choice to meet the above requirement
SANJAY BANGROO
19th February 2002 at 02:06 #11936NeilGuestGood info Sanjay!
I’m looking at a DTS system myself Ive heard these are good. Do you know who makes these?
Also I’m thinking about Avaya…but I dont know specifically what I need. I’ll have about 5 outbound callers calling on home consumers for roofing, painting and associated odd jobs. We will eventually grow to 15 in the next 18 months. What kind of Avaya phone system would I want? How does it compare to DTS? Are there any specific features I will want to look for?? I wanna stay lean!!!
Thank you!!
20th February 2002 at 05:40 #11937SANJAY BANGROOGuestFor outbound dialing u can go for an outbound dialer solution ( Hardware Based ) and u will not require any EPABX , AVAYA etc. since u are not having any inbound calls.
Go for DAVOX (now CONCERTO) or MOSAIC .
But these solutions are a bit expensive as such u can look for a local solution in your place.
You will find many such non- branded solutions available in the local market. Just evaluate them.I dont know where u r based .
Do let me know ?
I am in INDIA and we have many such home grown solutions readily available here.Further , if u have a customer database and are into some sort of CRM then both can be integrated also and u can have a SCREEN POP-UP too once the call is answered by your customer.
U can do without an AVAYA phone or the AVAYA switch .Just go for a non-properietary phone to save costs.In case u require any further info. feel free to write to me.
Regards and Best wishes
SANJAY BANGROO28th February 2002 at 04:56 #11938SANJAY BANGROOGuestGIRISH AND SHAILENDAR .
PLEASE GO THROUGH THE ABOVE ARTICLES AND MOST OF THE THINGS WILL BE CLEAR TO U.
IN CASE OF ANY FURTHER QUERIES REVERT BACK TO ME.SANJAY BANGROO
12th April 2002 at 09:42 #11939rajivGuestcan someone suggest me a call logging solurtion other than Nice logger
14th April 2002 at 14:00 #11940AnujGuestYou can look at TeleSynergy for the PABX solution
15th April 2002 at 09:12 #11941SANJAY BANGROOGuestGO FOR POWER CONNECT
7th May 2002 at 13:48 #11942Sanjay BhargavaGuestCan we run FR over IPLC?
The second question: How much overhead is going to be there? Passport 4460 mux from Nortel supports VoIP or VoFR and not TDM. Compared to using TDM mux like Kilomux, how much bandwidth is wasted if we go for VoFR on IPLC. Vendor say that 12.5Kbps per voice channel on FR. If we are buying a FR service from BT or others, this is fine as the overhead is not to our account and we will get the CIR committed to us. If we are deploying FR over our IPLC, the overhead of FR goes out of our bandwidth. Just like VoIP , VoFR has some overhead in carrying voice in pacet form. I think that this packet overhead is the extra 4.5 kbps on 8Kbps compressed voice. We will loose additional bandwidth for running FR on IPLC. Any idea bout it?8th May 2002 at 11:29 #11943SANJAY BANGROOGuestHi Sanjay,
I think u really have hit the nail on the head and without any doubt u are ABSOLUTELY correct. There are packet overheads in case of VOIP as well as FR. The only difference is that for VOIP the packet overhead size is around 7 Kbps and for FR it is around 2Kbps,per 8Kbps compressed voice.
PLEASE REMEMBER THE GOLDEN RULE :
‘FOR PURE VOICE NETWORK,TDM IS THE BEST SOLUTION’
Let me explain and compare various technologies in a small tabular form and this will explain everything to u very clearly :BANDWIDTH EFFICIENCY COMPARISON
(FOR A 64KBPS LINK )TDM VOFR VOIP
CODEC B/W 8KBPS 8KBPS 8KBPSPACKET 2KBPS 7KBPS
OVERHEADTOTAL B/W 8KBPS 10KBPS 15KBPS
LESS 6KBPS 9KBPS
SILENCENET AVG. 8KBPS 4KBPS 6KBPS
B/WMAX. 8 6 4
ACTIVE
VOICE
CHANNELSNETWORK DELAY AND JITTER
Key ingredient for good voice is
low delay
Delay can be measured as
Average length of time for packet to moveVariability in the arrival of packets
A jitter buffer nullifies the effect of variation
Latency is sum of above two
A latency of 200 – 250 msec makes a bad voice
Example consider network with 50 ms avvg delay but some part goes as high as 200-250 msec
A Jitter buffer of 10 msec will have low delay but will miss delayed packets
A jitter buffer of 200 msec will compensate for late packets but increase latecy
Hence Dynamic Jitter Buffering is a must criterion for Bandwidth ManagersCOMPARING TECHNOLOGIES
TDM
Quality of Voice : HIGHEST
Provides predictable and consitent quality of voice
Does not suffer from effects of Latency, JitterBandwidth Efficiency :
Very High for a pure voice network
Poor for a multiservice network
Cost : Cheapest
Effictive support for Multiservices: PoorFRAME RELAY
Quality of Voice :HIGH
Provides very good quality of voice under optimal conditions
Does not lower efficiency on implementing solutions for Latency, JitterBandwidth Efficiency :
High for a pure voice network
Very efficient for a voice/ data network
Cost :Medium
Effictive support for Multiservices: GoodVOIP
Quality of Voice : Optimal to poor
Provides very good quality of voice under optimal conditions
lowest efficiency on implementing solutions for Latency, JitterBandwidth Efficiency :
Very Low for a pure voice network
Not very efficient for a voice/ data network
Most efficient for data Networks
Cost : Medium
Effictive support for Multiservices: LowTECHNOLOGY FOR PURE VOICE NETWORK
Dynamic Bandwidth Allocation is not an issue
Voice Quality is of paramount importance
Bandwidth efficiency is significantly importantTDM is the best choice to meet the above requirement
TECHNOLOGY FOR VOICE /DATA NETWORK
Dynamic Bandwidth Allocation is an important issue
Voice Quality is of significant importance
Bandwidth efficiency is paramount importance important
FRAME RELAY is the best choice to meet the aboveDELAY OR LATENCY
Caused By:
algorithmic processing delay
frame delay (size of frame)
transmission delay
General Rule:
Delay approaching 32 ms requires echo cancellation
Delay is a function of transport technology alsoChoice of Technology has bearing on
1.Bandwidth Efficiency
2.Delay
3.JitterAND HENCE ON VOICE QUALITY
Choice of Technology is also governed by
Type of traffic running viz voice only or multiservices
Whether Dynamic Bandwidth allocation desired
QOS Requirement in multiservicesSOLUTIONS FOR JITTER
Variation in delay for Packet voice due to Network CongestionDoes not apply to Circuit Switched TDM
Happens when a large packet has already transmitted resulting in a voice packet waiting
Solved using techniques like fragmentation etc.BUT
EFFECTS OF FRAGMENTATION ON EFFICIENCY
Fragmentation results in creation of large no. of data packets
Each packet has to have a packet overhead
Packet overhead of FR is much lesser than IPIP overhead reduces WAN Efficiency by 10 – 15 %
FR overhead reduces by 2-4 %COMPARING PACKET TECHNOLOGIES ON QOS :
FR offers higher levels of prioritizationATM offers highest levels
IP is very limited in this aspect
TECHNOLOGY FOR PURE VOICE NETWORK
Dynamic Bandwidth Allocation is not an issue
Voice Quality is of paramount importance
Bandwidth efficiency is significantly importantTDM is the best choice to meet the above requirement
I hope I have answered all your queries. In case u require any further clarification or have more queries, please feel free to get in touch with me on sanjay_bangroo at Yaho.
Cheers!!!!!!!
Sanjay Bangroo8th May 2002 at 17:27 #11944SANJAY BANGROOGuestHi Mr. Bhargava ,
The table has not come out aligned properly after submission of the article.
In case u r not in a position to interpret the table ,
contact me on SANJAY_BANGROO@Yahoo.com , and I will send u the details ,Cheers !!!!!!
Sanjay -
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