Tag Archives: Telecommunication

SIP Weirdo Cheatsheet

9 Jun
A list of SIP protocol properties, rules and exceptions.
  1. From and To header fields of a registration request contain the same value.
  2. All the registration requests send from one UA to one registrar would always have the same Call-ID.
  3. As pre 3261(and 3261 alone) Invite is the only request which can initiate a dialog.
  4. Otherwise refer and subscribe requests can also start a Dialog.
  5. A calling UAC might receive any number of 1xx responses.
  6. An invite may or may not carry SDP.
  7. In case of forking, the calling UAC might receive multiple 1XX and 2xx.
  8. ACK is the only request without a response but still a complete transaction.
  9. ACK is a mandatory request in the same direction in which Invite was propagated.
  10. ACK is mandatory for an Invite, and doesn’t become part of any other transaction other than invite.
  11. ACK for all non-successful responses are considered within the Invite transaction.
    For more discussion on ACK visit : My Previous post on ACK
  12. Cancel request can’t be challenged for authentication, as these requests can’t be re-submitted.
  13. ACK and Cancel must have the same RR header field if RR was present in the Invite request.
  14. The request filed in CSEQ for ACK and CANCEL is always invite (3261)
  15. The CSEQ number of ACK and CANCEL is always equal to the CSEQ number of the INVITE request which they are bound with.
  16. Apart from INVITE (3261) the only other request which can be cancelled is an INFO request.
  17. As per 3261′s definition of transaction, the transaction is considered complete when a request is responded by any final response, still intermediate proxies would keep the transaction alive for a time equal to default transaction timer (if not configured otherwise).
  18. There can be no session without a Dialog, but this property is not orthogonal.
  19. you can not cancel the Invite transaction if at-least one 1xx response is not received.
  20. You can not cancel any invite transaction for which there are responses received other than 1xx.
  21. In any case (other than re-writing the complete Invite request) the To  and the From header fields never change in the entire life-time of a dialog.
  22. “z9hG4bk” is a weird string called as magic cookie. Nothing magical about it other than the name. It is used to by proxies to identify if the request complies to 3261 or its older brother 2543.
  23. 3261 doesn’t restrict 100 Trying for any request, so it can be practically issued for any request. Although it makes very little sense.
  24. According to 3261 and 3261 alone, Register is the only request which can be sent out of dialog.
  25. Option request is sent to query the far end of it’s capabilities.

slight deviation, Barging into other RFCs

  1. If offer is sent and no answer received another offer can’t be generated.
  2. if offer received but answer not sent, then another offer can’t be sent.
  3. SDP can be carried by Invite, 200 OK, 183, ACK none other.
  4. SDP answer must have equal number of m-lines.
  5. offer can be rejected but not ignored.

Enough for now, will keep updating this list.

One Bonus.
If worried about SIP with NAT read the rport RFC. Just google.


if there is any way in which this article can be improved, please let me know.

Thanks for stopping by.

Source: http://abhishekchattopadhyay.wordpress.com/2014/06/07/sip-weirdo-cheatsheet/


SIP Trunk BandWidth Calculation

9 Jun


Bandwidth as per Wikipedia is

A measurement of bit-rate of available or consumed data 
communication resources expressed in bits per second or 
multiples of it (bit/s, kbit/s, Mbit/s, Gbit/s, etc.). 
According to Hartley's law, the digital data rate limit 
(or channel capacity) of a physical communication link is 
proportional to its bandwidth in hertz.

Calculating the bandwidth requirement for a TDM network is easy. That is because they are based on either Multiplexing techniques like TDM,  Where each (Digital Signal) DS0 would correspond to one call, and depending on whether you are using T1 or E1 the number of simultaneous calls would be either 24 or 32 respectively. And at the end it boiled down to the number of connected circuits we have between two sites. However the same isn’t applicable for a SIP network when it comes to identifying Bandwidth, as there are no TDM or FDM techniques employed to send data. We’r going to use a century old telecom calculation technique called erlang. An erlang calculator can be found at http://www.erlang.com/calculator/ Prior to determining the bandwidth requirement of a SIP network we’d need to

  1. Determine, maximum simultaneous calls we need to support at any given time.
  2. Busy Hour Traffic (BHT): BHT is the measure of the call traffic at the busiest operational hour. also known as erlang load the Calculation ofBHT = (Average Call Duration(s) * calls per hour )/ 3600
    For example if we have 4000 calls per hour, with an 
    average duration of 180 seconds then BHT would be 
    --> (360 x 180)/ 3600 = 200 Erlang
  3. Determining Blocking: Blocking is the measure of failure of call attempts due to insufficient available resources. (as per definition number of lines).
    For example, a Blocking of 0.05 indicates 5 calls 
    blocked per 100 calls attempted. These blocked 
    calls would hear a busy signal or re-order tone.

The resultant of feeding these numbers in the erlang calculator is the number of trunks required to support the number of calls we wanted at a certain desired Grade of Service. Now if we had been working with TDM, then our calculation would have been complete with this resultant number from the erlang calculator. But since we are dealing with IP telephony and SIP there are a few more steps to be taken. In the next steps we would be converting that number of trunks (which is also equal to the simultaneous calls had it been TDM) into bandwidth. Lets see how.. For doing that we need to identify what codecs we would use. Whether we would use g711, g729 etc. Each codec has its own set of characteristics, which could include the codec’s sampling size, payload type, tolerance etc.. A short comparative list of capabilities of various codecs Codec Bandwidth Sample period                     Frame size                   Frames/ packet                Ethernet Bandwidth G.711 (PCM)              64 kbps                         20 ms                                160 1 95.2 kbps G.723.1A (ACELP) 5.3 kbps                         30 ms                               20 1 26.1 kbps G.723.1A (MP-MLQ) 6.4 kbps                    30 ms                                24 1 27.2 kbps G.726 (ADPCM)      32 kbps                          20 ms                                80 1 63.2 kbps G.728 (LD-CELP)   16 kbps                           2.5 ms                               5 4 78.4 kbps G.729A (CS-CELP) 8 kbps                            10 ms                                10 2 39.2 kbps AMR (ACELP)          4.75 kbps                     20 ms                                12 1 36.0 kbps AMR (ACELP)          7.4 kbps                        20 ms                                19 1 38.8 kbps AMR (ACELP)          12.2 kbps                      20 ms                                31 1 43.6 kbps AMR-WB/G.722.2(ACELP)6.6 kbps       20 ms                                17 1 38.0 kbps Before moving ahead we would also need to know the following

Codec Bit Rate (Kbps) Based on the codec, this is the number of bits per second that need to be transmitted to deliver a voice call. (codec bit rate = codec sample size / codec sample interval).
Codec Sample Size (Bytes) Based on the codec, this is the number of bytes captured by the Digital Signal Processor (DSP) at each codec sample interval. For example, the G.729 coder operates on sample intervals of 10 ms, corresponding to 10 bytes (80 bits) per sample at a bit rate of 8 Kbps. (codec bit rate = codec sample size / codec sample interval).
Codec Sample Interval (ms) sample interval at which the codec operates. For example, the G.729 coder operates on sample intervals of 10 ms, corresponding to 10 bytes (80 bits) per sample at a bit rate of 8 Kbps. (codec bit rate = codec sample size / codec sample interval).
MOS MOS is a system of grading the voice quality of telephone connections. With MOS, a wide range of listeners judge the quality of a voice sample on a scale of one (bad) to five (excellent). The scores are averaged to provide the MOS for the codec.
Voice Payload Size (Bytes) The voice payload size represents the number of bytes (or bits) that are filled into a packet. The voice payload size must be a multiple of the codec sample size. For example, G.729 packets can use 10, 20, 30, 40, 50, or 60 bytes of voice payload size.
Voice Payload Size (ms) The voice payload size can also be represented in terms of the codec samples. For example,a G.729 voice payload size of 20 ms (two 10 ms codec samples) represents a voice payload of 20 bytes [ (20 bytes * 8) / (20 ms) = 8 Kbps ]
PPS PPS represents the number of packets that need to be transmitted every second in order to deliver the codec bit rate. For example, for a G.729 call with voice payload size per packet of 20 bytes (160 bits), 50 packets need to be transmitted every second [50 pps = (8 Kbps) / (160 bits per packet) ]

Ok, The theater is all set now. Let’s jump into some calculations Bandwidth Calculation Total packet size = (L2 header) + (IP/UDP/RTP header) + (voice payload size) PPS = (codec bit rate) / (voice payload size) Bandwidth = total packet size * PPS Sample Calculation For example, the required bandwidth for a G.729 call (8 Kbps codec bit rate) with the default 20 bytes of voice payload is: Total packet size (bytes) = (MP header of 6 bytes) + ( compressed IP/UDP/RTP header of 2 bytes) + (voice payload of 20 bytes) = 28 bytes Total packet size (bits) = (28 bytes) * 8 bits per byte = 224 bits PPS = (8 Kbps codec bit rate) / (160 bits) = 50 pps => Note: 160 bits = 20 bytes (default voice payload) * 8 bits per byte Bandwidth per call = voice packet size (224 bits) * 50 pps = 11.2 Kbps Now let’s say we would have received a number 200 From the the Erlang calculator, then the required bandwidth requirement scales up to (bandwidth per call * total number of trunks that are needed) = 11.2 * 200 = 2240 kbps. On top of that there would be some percentage of additional (in a factor of 20-25%) bandwidth would be considered to compensate for factors like network re-transmissions, variance and collision. Which would eventually lead us to a figure of 28000 Kbps = 2.8 MegaBytes per second. So the bandwidth requirement would be an approximate 3MBps. However since each codec offers an entirely different sampling size it would be possible to achieve the same call rate with the desired GOM. Hope I was able to do some justice to the topic. If there is any way in which this article can be improved, please let me know. Thanks for stopping by. References: http://www.cisco.com/c/en/us/support/docs/voice/voice-quality/7934-bwidth-consume.html#formulae http://www.wekipedia.org http://www.cs.ru.ac.za/courses/honours/RTMM/software/52-VoIP-Bandwidth.pdf the image was taken from http://m.flikie.com/33582293/beautiful-night.html?cid=33554432&order=feellucky Source: http://abhishekchattopadhyay.wordpress.com/2014/06/07/sip-trunk-bandwidth-calculation/

Plotting the Telecoms Future

20 Aug

We are entering a major annual planning cycle for many players in the ICT industry. The challenge for the telecoms sector is that there are so many moving parts, some positive, some negative and all interacting with each other. And with the increasing digitisation of everything, factors from outside the telecoms market will seriously impact the future shape of telecoms. Indeed, one thought for the backburner is whether we will even think of telecoms services as a market in the future – but that’s for another day!

The basic equation for the telecoms market looks something like:

  • Demand from individuals, households and businesses and now ,‘Things’ (M2M) continues to grow as communications expands its horizons both in numerical and volume terms.
  • Total Revenues from traditional and new connectivity services are either in decline or about to go into decline – it is still a very big number overall, something like $1.4 trillion worldwide. The mix of legacy and new revenues varies by country but few doubt the gap left once the legacy services have washed through the system
  • New revenue streams such as TV/media, IT services, security, cloud, M2m all have attractive connectivity dependent components but they are also being addressed by other parts of the ICT industry and generally have lower margins than the connectivity services
  • Applications and content leveraging the telecoms networks are increasingly disconnected from the telecoms world and increasingly linked to the apps and links on our multiple screens through which we consume and execute
  • Maintaining a network infrastructure that can handle the explosion in traffic across all access methods and across the core, including in and out of data centres, needs major investment along with a rationalisation of the internal ICT infrastructure for most operators if margins are to be maintained, let alone grown

We need to recalibrate the expectations of the industry and its investors. Perhaps considering how many connections per household, individual, business and ‘thing’ require  and a fixed rate of revenue for each. This would define the worst case scenario, but still potentially very profitable. Add to this a percentage of the adjacent markets from ICT and Media and two-sided business models from pretty much every industry sector, and we have the potential future addressable market. However, remember that this new digital world means that the adjacent market incumbents can equally enter the telecoms space!.

There is fundamentally a lot of ‘spend’ at stake from all on the demand side. As everything digitises the demand side is increasingly likely to dictate through which channel the service (including connectivity) is consumed. So, a multi-channel strategy is needed along with major network and ICT rationalisation to bring the telco of the future into the new digital era.

Don’t get me wrong, Telecoms does have an underpinning role in the future scenario.  It may not necessarily be as the deliverer of the final service function or feature but there is a fundamental role at the heart of the new digital era for a trusted, reliable provider of the digital glue.

Source: http://chrislewisinsight.com/2013/08/19/plotting-the-telecoms-future/

Will Femtocell Eliminate Voip?

1 Oct

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A survey done a short while ago exposed which the advancement and possibilities for VoIP industry could access 32.4 billion market place worth of mobile VoIP by 2013. By 2019 it was predicted that 50 percent of all national and worldwide calls from mobile will likely be built by means of allIP networks. Along with the cheap connect with prices, a 5075% discounts when producing a call for lengthy distance or overseas calls employing portable VoIP telephone instead than 3G, GSM or CDMA cell phone managed to make it extra enticing for the buyers in comparison with the traditional telephones. VoIP is less costly as when compared with other telecommunication programs this kind of as PSTN, GSM, and CDMA mobile phone among others. The long length countrywide and international calls can virtually be absolutely free or endless with its least expensive plans, discounts and promos in comparison to the typical cellphone choices for around 10 to twenty instances less expensive. A call from a cell phone through VoIP technology lets absolutely free incoming calls in many nations around the world.

Alright, let’s return somewhat and study Femtocells vs POTS. Aside from the proven fact that POTS carriers are usually the same fellas who very own the cell corporations and also have zero fascination in competing with by themselves you’ll find however a couple of things you just are unable to do with femtocells. Particularly at any time attempt to mail fax by way of VoIP? Should you have, you most likely are aware that it’s really a very difficult activity despite the most effective of settings. Now let’s go a step further and think about seeking to fax through your cell phone. Sounds preposterous? yep, it positive is. Fax and modem more than VoIP will most likely function perfectly sooner or later, but I just will not see mobile phone makers deploying T.38 faxready handsets each time quickly.

A web based cellular telephone directory that permits you reverse lookup to be able to determine not known mobile phone quantities and lookup of unlisted, private, and unpublished phones numbers can make quick function of your respective directory analysis. You are going to have the ability to have in contact with your extended misplaced friends and kinfolk like a high quality on the web directory presents many of the details in the user for virtually every landline or mobile amount inside of a fraction of seconds. This revolutionary style of registry can be utilized to start looking up cellular telephone quantities as a way to trace prank or harassment callers.

Source: http://www.iccup.com/dota/content/blogs/will_femtocell_eliminate_voip.html

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