Category Archives: Technical

Which Battery Cell?

Batteries

Need for battery cells seems endless; there is always something running out of charge.

Which one to buy? Here is a small introduction (mainly focusing on most common types of disposable batteries available):

Alkaline Batteries Lithium Batteries Zinc Cadmium
Lasts for Less Time Lasts Much Longer Original & Old Technology
Small Shelf Life, Approx 5 Years Longer Shelf Life, Approx 10 years + Carries Least Capacity
Cheaper Expensive Cheapest
Less Suitable for Harsh Weather Better Performer in Cold & Hot Weather
Environmentally Harmful if not disposed off properly
Good for Low Drain Devices e.g. Remote Control, Clock, etc Good for High Drain Devices e.g. Toys, Torches, Mechanical things
Heavier Lightweight (1/3rd of Alkaline)

So now you know which one to buy 🙂

Here is a Link if you need more information (external site): http://michaelbluejay.com/batteries/

Boeing 787 Extreme Take Off

787 in flight
Boeing 787 Dreamliner

Aeroplanes are the No 1 human invention in my view. It just amazes me beyond dreams when I see a plane taking off but this extreme Boeing 787 take-off will blow your mind away. It was conducted in an air show in Paris this year (link below to download & play).

Some Fact about Boeing 787

787 Cockpit
787 Cockpit View
  • Boeing 787 known as Dreamliner was first rolled out in 2007.
  • First flight took off in December 2009 and first commercial flight took off in October 2011
  • There are around 300 planes in service currently.
  • Boeing 787 can accommodate up to 335 passengers.
  • Its wings are elegantly curved upward such that it appears like a bird floating in air. This design feature reduces air friction as well noise, making it more fuel efficient.
  • 787’s list price starts from $ 146 million for basic model, going to up to $2 00 Million
  • It is a long range, highly modern, two-engine passenger plane.
  • most of its systems such as braking, pressurising, de-icing etc are all electrical.
787 Take Off
Near Vertical Take Off

Link to Extreme Take Off Video (Download & Watch): https://www.dropbox.com/s/v5qh8olae2neskq/Boeing%20787%20Dreamliner%20Extreme%20Take%20Off%202015.mp4?dl=0

Session Initiation Protocols for Voice over Packet based IP Networks

Introduction: The Need for Sessions:

Telecom industry has rapidly adopted IP for its voice & data networks. IP protocols are excellent for data services but are not so suitable for voice, especially when it comes to setting up, modifying or closing a real time session (or call) which is the pre-requisite for a voice call. This function is performed by ISUP (an SS7 Signaling Protocol) in traditional telecom networks.

Examples of available IP based session control protocol include SIP, SIP-I, SIP–T & BICC, which have the capability to establish packet switched based voice calls, hence are used or considered for session control in PSTN/GSM/3G/4G. It is desirable for the industry to adopt a single protocol for a convenient packet based voice (VoIP) development & deployment.

BICC: Bearer Independent Call Control was standardized by ITU-T in 2000 called Q1901 – Capability Set 1 (or CS1) to provide call control functionality. Later in 2001, BICC CS2 was introduced with BCP (Bearer Control Protocol) to provide IP Network Bearer Control & CODEC control. There is no further development after that (why?? Read below). BICC inherits the ISUP messages & parameters, hence providing the same services & allowing natural interworking between ISUP & BICC. It was adopted by 3GPP for UMTS in Rel 4 (2001).

SIP (Session Initiation Protocol) was developed by IETF (RFC 3261). It is most commonly used in the IP world. IETF neither intended nor attempted to make it suitable for PSTN. It had 3 limitations from voice session point of view:

1-      No Direct Interworking with PSTN (more precisely ISUP SS7 Signaling).

2-      SIP session model is StateLess (no message sequence) whereas PSTN ISUP call model is strictly StateFul i.e. Sequence of messages is important ( e.g. IAM -> ACM -> ANM -> REL -> RLC).

3-      PSTN ISUP based features cannot be implemented in SIP because most of that functionality is not available in SIP.

Later on IETF & 3GPP (including ITU) realized that SIP to ISUP interworking is necessary for voice over packet switch (or more precisely IP) network. But two parties, IETF & 3GPP/ITU continued independently developing methods & ways of encapsulating ISUP messages in SIP. The result was two SIP variants as mentioned below:

SIP-T (SIP for Telephony): IETF defined SIP-T in 2002 (RFC 3372, 3393, etc). It provides mechanism of carrying ISUP messages as an attachment to SIP messages.

SIP-I (SIP for ISUP): ITU developed & standardized SIP-I (Q1912.5 Profile C) signaling protocol allowing complete interworking between next generation IP networks & traditional PSTN (or GSM/UMTS) networks. It helped extending the services that can be offered in the VoIP domain. SIP-I encapsulated ISUP messages. Because of its flexibility, 3GPP Mobile networks (3G, 4G) adopted SIP-I as an alternative to BICC.  SIP-I was formally adopted by 3GPP for IMS.

Why BICC Dropped in Favor of SIP (I & T)

There are three reasons why SIP is preferred to BICC:

1- There is no further enhancement work done in BICC at any forum making its capabilities stagnant.

2- BICC was intended for voice in GSM/UMTS, hence it is limited to serve these only. When it comes to establishing session for multimedia systems, BICC has nothing to offer.

3- BICC uses specific (proprietary) framing protocol IuFP. It is not widely deployed compared to SIP framing protocol (based on IETF). Even ITU version of BICC uses IETF based framing.

COMPARISON: SIP-I & SIP-T)

Comparing SIP-T (from IEFT) & SIP-I (from ITU) suitability for voice, SIP-I is more suitable for GSM/UMTS/LTE environment due to the following (except pt. 6):

1-      ITU has more experience, relevance & future aspirations for voice compared to IETF

2-      Trust Zones & Security Environment: SIP-T assumes there is no trusted zone hence more complex. SIP-I is opposite to it.

3-      Encapsulation Procedures & Message Mapping: SIP-T is under-specified (under developed) hence still not fully trustworthy for deployment.

4-      Support for RFCs & Standards: SIP-I has more developed RFCs & standards.

5-      User Plane IOT: SIP-T does not emphasize on user plane interoperability testing.

6-      Forking: SIP-T supports Forking feature, the ability to create mid-call multiple streams (or branches) of audio associated with a single call and then send those streams of data to different destinations. It allows service providers to use technologies such as speech recognition, voice authentication, and text-to-speech conversion to provide sophisticated services to their end-user customers. SIP-I does not support this as yet.

Hence there is more uncertainty & an increased risk in deploying SIP-T. SIP-I is more suitable for the converged GSM/UMTS/LTE/NGN/IP networks. SIP-I is adopted by both 3GPP & ANSI. Moreover 3GPP plans to incorporate SIP-I in specifications.

SIP is Layer 5:

SIP ( I & T ) are layer 5 protocol. Hence they need something at layer 4 to work with IP (layer 3). UDP & TCP are available but more reliable is TCP.

No

Layer

Protocol

SIP ov IP

Protocol

SS7 ov IP

Protocol

BICC ov IP

     

ISUP

 
5 Session

SIP – I , T

SIP – I , T

BICC

4 Transport

TCP / UDP

TCP / UDP / SCTP

SCTP

3 Network

IP

IP

IP

2 Data      
1 Physical      

 

So Which one to Use?

BICC, SCTP, SIP-T or SIP-I are the protocol choices to setup a voice call and carry SIGNTRAN (SS7 Signaling over IP).

The above discussion shows that SIP is a good candidate for voice call setup in IP networks. And SIP-I is the preferred choice for Voice networks to carry Signaling over IP.

NOTE: Above discussion is based on OSI 7 layer model. IP has its own 4 layer model. Its concept is similar but layer numbering may differ. 

The end.

Why SCTP (Stream Control Transmission Protocol)

New Trends & Protocols in Voice Network:

Understanding Transport Layer (4) Protocols is vital in understanding how data is transmitted from one device to another successfully. Traditionally these protocols were used for data transfer. Last decade has seen a new trend emerging of sending voice over PS (Packet Switched) networks opposed to using traditional CS (Circuit Switched) networks. VoIP (voice over IP) is one such prime example. Leaving merits & demerits of Voice over CS or PS aside, we will focus on new protocols emerging in this area, especially SCTP. Let’s start with few important basics:

IETF: The Internet Engineering Task Force is an international body based in USA working on protocols and development work relating to internet.

ITU: international Telecommunication Union, based in Europe, controls developments and protocols / standards relating to communication, primarily for voice.

3GPP: Part of ITU, it focuses on GSM, 2G, 3G and 4G development and protocols.         

Layer 3 Protocol: IP Internet Protocol is a connection-less one, and hence lacks reliability. To supplement, Reliability is provided by layer 4 protocols mentioned below.

Layer 4 Protocols: TCP / UDP / SCTP (Stream Control Transmission Protocol): These are all transport layer (4) protocols. They are responsible for end to end reliable delivery of data. All are connection oriented, i.e. need connection or session association with pier before working (otherwise who will they ask for acknowledgement??)

Read detailed Discussion on TCP & UDP Here.  

UDP & TCP were designed for Data (bits):

UDP is packet (chunks of data) oriented. A fixed sized data chunks received from the application and then forwarded in UDP packets. UDP is un-acknowledged (may be one acknowledgement after 10 packs). It is light weight and hence used where good speed is important than reliability, e.g. voice.

TCP is bytes or stream oriented. A stream of byes of any size received and then divided into TCP packets and sent. TCP is made more reliable using Acknowledgements. So it is good for file transfer because file transfer does not tolerate any loss.

 So Why SCTP? It is a relatively new protocol developed in 2000. It was designed as Message Oriented (opposed to bit oriented). It gives importance to the order of messages also. Hence it is suitable for message oriented applications, e.g. SS7 Signaling. It also considers session/path loss, which is required for a voice connection. In short SCTP was developed specifically for carrying signaling in VoIP applications.

Who Did it? SCTP was designed by IETF’s subcommittee responsible for SIGTRAN (SS7 Signaling Transport) Protocols. IETF is more IP & internet oriented & has experience in this area, and hence bias towards this area.

Problems: The fact that IETF developed SCTP created two problems:

1-    IETF lacked experience in voice Signaling. SS7 Singling protocols were produced by ITU. So SS7 & SCTP did not fit well.

2-    IETF lacked experience in Voice networks. Therefore SCTP was not liked (preferred) by 3GPP/ITU but they were in need of something in this area.

There was a need for something to carry SS7 Signaling over IP as VoIP was getting popular. IP & voice networks were getting merged.

So what was the Solution, Read it Here:

Iran Telecommunication Market Analysis (2010)

I visited Iran in 2010 (Read my story Here). Below is a summary of my findings of the Telecom market there:

Demographics:

 Total Population:    74 Million

Urban Population:  60%

Young Population:  36%

GSM penetration:  60%, expected to grow up to max 80%.

Internet usage:      36%

Fixed Line Penetration:    High in urban and low in rural hilly & desert areas.

Fixed Line Operators

There is only one fixed line operator TCI (Telecommunications Company of Iran – state owned). Its switches are old PSTN based. There are plans to introduce NGN within next few years.

 MNP (Mobile Number Portability) is not introduced but may come in next 5 years plan.

 Wireless Operators:

 There are two 2G licenses held by MCCI & Iran Cell.

A 3G license for 15 million subscribers has been issued to a new entrant ‘Tamin’ for 15 years, with 2 years exclusivity (no one else can have 3G for two years at least). This ignited the market. Competition is expected to rise with the entry of 3rd operator. MCCI having the biggest customer share is very worried of losing its market share. Radio coverage is little patchy outside cities and network performance is poor till now (2010). But operators are feeling the need to upgrade their networks to retain their market share.

This has also activated the mobile network vendors. Huawei, NSN and Ericsson have been quite active. 

Three WiMax licenses are also issued in 2010.

1. MCCI (Mobile Communication Company Iran): Started in 1994, and is owned by TCI (Telecommunication Company of Iran, only fixed line state owned operator). Subscribers are 32 million with 60% market share. 60% subscribers are postpaid. Nearly all business users are with them and cannot leave because of MNP not being offered in Iran. Moving would mean losing years’ old number, which is no desirable for customers.

Core Suppliers:     Nokia 70%, Ericsson 20%, Huawei 10% (PS & little bit CS)

Radio Suppliers:    Nokia 60%, Ericsson 40%

2. MTN IranCell: Started in 2006. 15 million Subscribers with market share of 35%. 98% are Prepaid. It was the first one to offer prepaid so most prepaid subscribers joined it. There is a potential for MTN Iran in post paid area when MNP is introduced.

MTN has WiMax network which is for data only initially.

It has 2G & 2.5G (EDGE in limited areas). Data usage is low with subscribers less than 10%.

MTN Iran is planning for IMS deployment.

Core Suppliers:    Huawei 100% (ASN gateway is also provided)

Radio Suppliers:  Nokia 40%, Ericsson 30%, Huawei 30%

3. Remaining 5% mobile market share goes to 3 small regional operators.

The end of document….