Leader’s Talking Nonsense

Sometimes the communication skills of Leaders & Politicians really surprise me. It is amazing how they can manage to still say something positive  (also known as bull sh**t)  even during disasters.

Alcatel-Lucent CEO and Members of the Board stepped down (in 2008) due to such a bad performance that the newly merged (Alcatel & Lucent) company’s shares went down by more than 60% in the last year, and the net Loss equivalent to $1.7 Billion,  was larger than analysts’ expectations (they were shocked basically). 

The news reads:

“Alcatel-Lucent, one of the world’s largest telecommunications manufacturer,  posted a second-quarter net loss of 1.1 billion Euros and announced that its  CEO, Patricia F. Russo, and its Chairman Serge Tchuruk, would step down.

 Now  look what the Leaving CEO and Chairman Stated:

CEO Ms. Russo said “she was Pleased with the Progress the Company was making, but that it was the right time to step down. The company will benefit from new leadership aligned with a newly composed board to bring a fresh and independent perspective that will take Alcatel-Lucent to its next level of growth and development.”

Mr. Tchuruk  the Chairman said . “I am proud that Alcatel-Lucent has become a world leader in a technology which is transforming our society.”

WOW What a performance!!!!!!

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.


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.






SS7 ov IP





5 Session

SIP – I , T

SIP – I , T


4 Transport




3 Network




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:

Mr Rehman Malik says Wives & Girl Friends Kill in Karachi

About 200 people are killed in Karachi during last few days’ violence due to rivalry in political parties.

Interior minister of Pakistan Mr. Rehman Malik pronounced that “out of 100, only 30 were victims of targeted violence in Karachi while 70 others died at the hands of their wives or girlfriends.”

I feel like laughing first and then crying. I don’t know what to do? Please help



Click to enlarge the cutting above

Funny Quoutes

  1. Do not argue with an idiot. He will drag you down to his level and beat you with experience.
  2. Better to remain silent and be considered a fool, than to speak and confirm it.
  3. If you can’t impress people with your intelligence, confuse them with your bullsh1t
  4. The last thing I want to do is hurt you. But it’s still on the list.
  5. If I agree with you we’d both be wrong.
  6. The early bird might get the worm, but the second mouse gets the cheese.
  7. A computer once beat me at chess, but it was no match for me at kick boxing.
  8. Never get into fights with ugly people, they have nothing to lose.
  9. Worrying does work! 90% of the things I worry about never happen.
  10. 10.  My psychiatrist told me I was crazy and I said I want a second opinion. He said okay, you’re ugly too.
  11. Some cause happiness wherever they go. Others whenever they go.
  12.  I like work. It fascinates me. I sit and look at it for hours.
  13.  I used to be indecisive. Now I’m not sure.
  14.  Knowledge is power, and power corrupts. So study hard and be evil
  15.  A TV can insult your intelligence, but nothing rubs it in like a computer.
  16.  Going to Mosque doesn’t make one a Muslim any more than standing in a garage makes you a Mechanic.
  17.  Knowledge is knowing a tomato is a fruit; Wisdom is not putting it in a fruit salad.