 So, we will go ahead with whatever I was describing yesterday and now basically we have to look at the what will be the elements which will be participating in a view IP system. So, first of all let us talk about voice over IP, we will expand this thing on to other kind of media streams also parallely. I think another one which is I know because I have designed that system is and I have classrooms over internet for example, where millions of students or trillions of students can actually attend a lecture. Now, if you can even build up that kind of system, but it is not only audio video, nearly two-party audio video it is possible, multi-party audio video is also possible and there is possibility of actually using live chat, there is possibility of tracking how many people are there, there is to be somebody who will be master who will be controlling and people will join the session, they will leave the session, they will raise their hand, they will ask questions, there I might actually conduct a live poll in a classroom, I might give a demonstration or a lot of interactive activity which is possible. Now, this is all together different kind of media streams which are viable. Now, when the SIP or the session initiation protocol was designed, it was you have to design a protocol, now forget what is SIP, suppose we have to design a session basically a mechanism by which any kind of this media session or interaction session can be created among any number of people. So, the idea is basically how this can be created, I need to build up an infrastructure through which this kind of setups, this kind of sessions can be identified whether they are existing or not existing and then people can join these sessions. So, as far as this whole infrastructure is concerned is only for initiating you into the session, once you are joining the session you are into that, then this protocol or this mechanism may not work, there might be some other protocol to which you will be handed over. Very simple example is a conference call where there are 10 people who are in a conference and these 10 people when the conference usually done, there will be a master who will be always controlling the conference, he will say now you can talk or you can talk somebody who will be a floor manager, so floor management system is not part of the design for SIP, through SIP you will be joining this thing and SIP will then inform you this is my floor management mechanism, there will be a separate protocol for that. Now, when you will talk you will be using some kind of audio streaming live audio streaming, if it is a video call may be video streaming also, but then what kind of video thing will be encoded or transmitted at what rate, what will happen if there is a congestion, SIP does not bother about it, SIP is, so SIP will only make sure that people join the session appropriately and after that the two peers will negotiate with each other and will do whatever they want and they will communicate. So, technically what we are doing is we are now building up a peer to peer system, so telephony through peer to peer, but this is fine in a futuristic system, SIP is a peer to peer, when two SIP clients are there they talk, they communicate directly into end, but usually there will be problems in the network, so sometimes peer to peer technically is not possible. So, it requires sometimes what you call intermediaries, which have to be because of the network issues like using private IP blocks or when you are behind a proxy or when you are behind a net, so it becomes a problem. So, it requires some intermediary nodes, so we call them reflector in some of the designs, some of the design these are known as super peers or super nodes, some places these are known as proxies and so, but these do exist. Now, what will be the elements? The whole world will not be view IP in Monday, you have to coexist with the existing PSTN and PSTN telephony also has to be connected, that is another requirement. So, you will have a world where we will have a internet, we will have voice over IP clients, so I am now moving one step further and I am now including my PSTN network also into the system and I will also have a PSTN network, PSTN is public switch telephony network, is a conventional circuit switch system in which our telephones are actually connected and you also will have GSM, you might actually have, see do you remember these are also separate networks, they actually override, so this blob might be partitioned two parts interconnected through a conventional PSTN, but this actually technically is a separate network, but it does gate weighing with the PSTN, that is why your landline can call to the mobile phone. Now, this kind of blob is also going to come into the picture and good thing is that this requires only internet protocol, so whether you are using in the lower layer a Y max or you are using 3G does not matter, so far it provides IP connectivity I can use wipe system. Now, how you will connect to the PSTN, I am not bothered about these, so far if I can connect to PSTN I can connect to all these that is my assumption, so there is no direct peering requirement between this and this, this also can be done, but the same device which I am going to put here can actually do this job, so I need a gateway here, this is technically known as media gateway, so when you look from this side all these phones will look like their wipe phones, when you look from this sides all these soft clients, we call them soft client because most of the wipe telephony clients are nothing but software, they can be put on embedded system and we call it a wipe phone, you can actually buy these phones and most of our actually all smart phones have a built in capability of becoming a wipe phone for example, it is just by loading a software. So, this media gateway does this translation activity, now one way is that whatever be the number of phones on this side all of them can be mapped here as wipe phones and I can run multiple instances for signaling for doing peering bit for voice transport or video transport between any of these phone to here and then there the translation will be done and of course, there will be limited capability if it is a conventional PSTN only voice will be there, you might be using a packet switch system all the way for transporting your voice, you might be using a different codec, when you come on to this side this circuit switch system you will be transmitting raw analog audio it is possible or if this can be that 64 kbps what we call this audio which is there TDM stream that can be transmitted then converted to analog at the edge exchange and then your phone can be connected to this, but usually these devices are costly. So, in fact people usually want their own conventional phone because they always are habitual with that even to use that kind of thing some people might connect a small box this is also known as media gateway and then in turn connect their conventional phone on to this, this is also pretty popular system. So, you actually feel as if you are on having a conventional phone you still dial number and everything, but whatever you do this box captures all the signaling it has a built in processor and it does all the signaling everything which is required by wipe and can connect to outside world actually. In fact, there have been a pretty popular system I have actually seen that in companies who is calling from here to United States is pretty costlier if you actually use a PSTN phone. Of course, now it is that cheap the reason is because most of the operators take the call from your PSTN line they actually convert it into a via wipe call and through a media gateway they will transport all the way to the United States or whichever country you want and from there they will again convert into a through a media gateway into their PSTN network. So, long haul communication is not through circuit switching it is through packet switching. So, efficiency is pretty high and call cost can be very small, but interesting thing is that I do not know whether you have seen I have seen this in India people whose relatives are living in United States they usually send them a small box. So, there are many companies will sell them and it is basically a box in which on the one side you can connect the phone it is a media gateway. You bring that box attach in your ADSL modem or Ethernet and once that is done only thing you should not have a proxy in between or netting router should not be netting router is fine, but proxy should not be there in between because it is working over a UDP tunneling or TCP proxy, but if it is STTP based transport is not supported in that box, but now the more advanced boxes may be supporting that. So, even it can go behind a proxy if you can configure proxy user and proxy password. So, what happens the moment you connect this phone this phone the moment that box gets activated will try to register to a server that indexing server of the operator in United States and it is IP connection. So, it can register there it registers there in that registry it will register this particular address as this particular IP address of this box as nothing, but what we call to a phone number. So, the phone which you are having in India actually is having a number which corresponds to US number. So, in US people will think it is a local number, but actually the phone number lies in India the call is routed through WIPE because register for this phone is sitting in United States. So, it has given a different number, but you need not have phone number as I told last time yesterday class it has to be some user idea at the rate domain name. So, if the Vonage is providing this thing I think this is the one company which does that. So, this prefix is automatically added by that phone and then there is a number which is US number which is your choice when you buy that box. Now, signaling people have built up proprietary. So, let us come to the features of the signaling and what all will be required. So, today what I have added is this one extra element which was not there earlier the media gateway. So, I have to handle this media gateway I have to handle these clients through this media gateway I have to handle the soft clients. So, Skype also when you are making a call actually is exactly doing same thing Skype talks through their Skype network with Skype clients there is a media gateway which Skype has put at lot of places. In India I think it is in US they have, but I think India they do not have a gateway because when I dial from Skype client to India I do not get Indian telephone number. So, only problem is the caller ID in that case. So, when you do a dial you dial with a Skype to a phone number you can do it actually if you have bought the credit. So, once you do that so technically this media gateway is running a Skype client. Now, this remember is a proprietary system which must have been built by Skype. Most of the media gateways come on this side with the signaling interface which is either H dot 323 or SIP there are only two standards which are very commonly used. SIP also has two versions what we will be talking about is this RFC 3261. This is ITUT standard and mostly is kind of it is still in use, but most of devices people want only this one this is a open standard by IETF dot ORG. Now, this SIP v2 is only handling session initiation or session creation basically and joining the session process and how you will get to know that to whom to contact. For example, what is your phone number how do you know your friend's phone number that friend must have given it to you or you must have found it on a website or it must have come in his mail thing the bottom of the mail. It is the same way these URI's SIP URI's we call SIP Universal Resource Identifier. So, phone number the equivalent is known as SIP URI. URI stands for Universal Resource Identifier SIP stands for session initiation protocol. Now, this is not the only thing there are many other set of protocols which are required along with this. So, once I will come to this how the SIP actually handles the session setup. Once the session setup is done this client will be talking directly to this usually. Usually in most common practices this will be done in this way, but also there is a requirement for example, tomorrow if this SIP server or that indexing server or register will be managed by say BSNL and Tata telecom or reliance everybody gets a license. So far it is not there, but may be in a half year down the line we will actually have those systems running in India also. Currently it is not permitted. What is permitted in India is only this kind of system. Only thing which is only for call centers I think it is permitted to actually have a wipe system from conventional telephony to wipe conversion that is permitted because they run call centers. For example, for US if you are running a call center this is a US number on which the people are going to make the call. And that call actually is all they were routed over internet all the way to India. So, those numbers which is their panel actually is a US number which is there, but Indian operators just provide them connectivity and they have to actually keep on giving the reports. One of the biggest problem in this case is you cannot tap the signals. If two peers connect directly you cannot tap. If they both agree on some encryption algorithm and the payload itself is encrypted even if you tap the packets you cannot figure out what is there inside. Skype is I think is a pain in enact problem for Indian defense as well as home ministry everywhere because they cannot tap those things. And you might have heard that during that Bombay stuff they were actually using Skype and nobody was knowing what they are talking. And somebody from across the border was monitoring through Skype only. So, it is still a big problem Skype trapping unless Skype company itself basically cooperates. But probably now it is viable because Skype is taken over by Microsoft. Microsoft has an Indian office. So, Indian government can throttle them and make sure they cooperate. Earlier it was not possible Skype was not a company in India. There was no incorporation here, but now it is. So, I do not know what is the current situation. So, home ministry must be taking care of that. Tapping is not possible. So, you require a different kind of mechanism. This is usually not announced, but when you are actually put a tender whenever you buy an equipment it is a mandatory condition and it is advisable that you should put it. You never know when you will you yourself may require it or somebody else in the government may ask for it actually this particular scenario. We require what we call intermediary gateways. This is again a special kind of media gateways nothing great. So, what will happen is I will give an example. For the time being forget SIP server. So, there is an indexing server running. I call it. These are more generic name actually indexing servers. So, most of the peer to peer systems will have either a distributed or a centralized indexing. So, I am assuming it to be centralized indexing server here. So, when they actually this guy I want to talk to this person. Assume is like a Skype kind of client or Google chat kind of client. It will talk to this person and say I want to talk to so and so people. So, everybody updates their status here. So, when this person wants to talk to this guy. So, he will be this guy will inform him that somebody wants to talk and if he agrees to talk to this person. So, it will transfer its IP address to this person. Its IP address will be transferred to this guy. So, they both know each other and then they will start talking to each other. So, the security somebody else will not be able to spoof because the port numbers which are assigned for every communication they are dynamically changing. So, which port number will be used for communication is assigned by this guy register. Once those are done they can just keep on communicating on that basis and can boot a step for more media streams. But this is dynamic this there is no static things. So, they have to keep on informing the port number at which this is active to the register or indexing server. Now, if I want to implement a tapping system how it will be done I want to tap a certain number. So, I require then additional server the way it will be done is this indexing server will know its IP address this knows this IP address. This will also talk to this third guy for media gateway which can create duplicated entities which can create kind of a conference call. Every packet which is passing through this filter is duplicated and can be rooted to something third party. So, usually what will happen is this guy will be informed IP address of this person this guy will be informed and another port number and IP address of this. So, this guy actually will see this IP address and port number is x and this guy will see z as this IP address and port number. So, when they will be talking this media will be passing through this intermediary. An intermediary can be directly controlled not by indexing server there will be a command which has to be given by your NOG or the operators command center and they will control this gateway and they can make duplicates of all the packets and can root to another client where the call actually can be listened parallely or it can be recorded it can be tapped actually this is what is the tapping principle. In conventional telephony also tapping does not mean that you go physically and put your wire and listen what the other persons are talking about this is not the way it is done. So, once you just inform when the call is being rooted through a switch it is just being a copy gets created and copy is also rooted to another number through again a switching path. So, that is how in Delhi there is actually joined wing of three forces which mean to which actually does this particular job all calls can be rooted to them without any issues and from any switch in India. So, these are just remotely configured from the NOG and any call which is passing through they just make a duplication it is rooted. So, this either they do it or I think the pull is actually does it in some cases when the call tapping is required and of course, one category is doing through mobile phone mobile phone as a weak encryption. So, if you are in the same cell where the transmission is happening you can listen to those slots in that is another way of doing it, but you need to know that how the call is from which number to which number at least one of them need to be known. So, this is the tapping mechanism which will be implemented. So, usually these kind of media gateways again need to be controlled and these are not done through SIP. SIP only initiates in sets of the sizing. So, for doing this media gateway control you need to do something more only SIP cannot do this. So, lot of other kind of command sets are required and we use a protocol called mega code for this. So, complete system is not SIP complete system is many more things actually I will just list most of them which are used. Then once we actually are doing a peer to peer communication between two end points the voice has to be transported it is a real time streaming. So, at time when you are playing certain things has to be fixed. So, time gap at which you have taken the samples and the recording side it has to be played back in a similar fashion on the receiver side. So, this requires a real time transport protocol this is again independent of SIP. SIP does not bother about it actually how the media is handled. So, you will require also RTP and I think I have there RFC numbers which are there. So, this is RFC 1889 3015. Now, RTP is for real time communication or control basically, but for the streaming media where packet loss is once a while it is loss is loss does not matter you do not require reliability you require real time streaming protocol this is usually for media audio and video not for the control signals RTP is for control signals. So, RTSP real time streaming protocol will also be part of the design this one requires an RFC of 2326. Now, when we will actually use SIP authorization authentication everything will be taken care of the two peers can talk, but what SIP actually does is provides a mechanism by which a object we call it OPEC object SIP never looks into what is there inside that. So, it is nothing but a file this file is simply transported to the right person he also sends back the file in the bag based on this this guys will share the information. So, they will share what is my IP address what is my port number at which you should connect. Honestly speaking this indexing server does not tell in SIP case once the SIP call is set up these guys through this route will inform that this is my IP address and port at which you should connect this guy will also tell this is my IP address and port at which you should connect. So, all the signaling will be routed through indexing server or SIP server we call it SIP proxy that is the term. So, that format of the file which will be transported through SIP is known as session description it is not technically it is not a protocol it only gives the format of the file through which a session can be described any generic session can be described is a format of the file is a text format it is not XML schema. Nowadays of course, everything we tend to use XML schemas because they are more generic, but this is a purely text based system which was built or evolved over the time actually and we call it SDP the term is session description protocol, but it is not a protocol as such and this one is 2327. So, system which we are going to actually have another two months in IITK will actually be based on SIP v2. We will have our own media gateway huge number 6000 analog lines will be terminating here. So, there is no exchange there is no switch on this side remember this I am talking about PSTN. So, there will be conventional switches in our design from here itself we will actually have 5000 wires going all the way to the each individual phone which is there. So, battery power will also be given by this equipment conventional telephony battery power is provided by exchange, but why telephony it is not provided by the exchange? It can be there is I triple this 802.13 I think that is what 12 one of those standard is there power over ethernet POE we call it. So, if you are most of the time what will happen is how you will connect your wife phone to the IP network most of the time it will be ethernet port which will be used you can actually use Wi-Fi nobody stops you can use Wi-Mex nobody stops you can have any other kind of physical layer or data link layer you can actually use on top of it. So, far it provides sufficient bandwidth in fact when negotiation of the codec will be done through this SDP thing which will be transported through SIP mechanism that time itself bandwidth constant will be taken care of. So, depending on available bandwidth different codecs might be chosen well a Skype also does the same thing. So, if the bandwidth is different Skype dynamically changes the codecs and that is why the quality of voice is almost it is not degrading much, but it degrades if the bandwidth goes down. Most of the time these are embedded devices. See, when you buy a mobile phone it contains I think most of them are built either with some processor. So, ARM has been kind of become a de facto standard, but of course now ATOM is also being used extensively for smartphones. So, in India we have got I think only one manufacturer providing a phone with ATOM most of the other smartphones are all with ARM series, but these are pretty much very powerful processors. So, it is like a computer your android phone is a computer technically. We will be actually buying again a smartphone kind of stuff we have already bought them, but it is a desktop versions it is a pretty much like a computer actually it comes from there is, but there is no keyboard keyboard is keyboard is like a telephone keyboard. The box will look like a telephone and basically it comes into that and they do not identify a connection. Yes, yes right, right it is exactly the same thing it will be connection to Ethernet port, but conventional phones also can be used by putting a small media gateway box. So, go to Cisco dot com dot com site the many manufacturers which are making this small box a small dub bar. It also has a small processor which is sufficient enough to do encoding, packetization, transmission, chip signaling your RTP there is no mega core required in this case this media gateway is not controlled by anybody. This is uncontrolled media gateway is only for one port. So, it is a different name given, but technically it is a media gateway, but does not require a mega core protocol. In fact, you can do away with mega core you would need not actually have it you can still do SIP based control, but then you require lot of power here which need to be done, but if you are connecting to a PSTN in our case SIP actually was fine, because we are not having actually any switch our condition was there is a media gateway and then there are 5000 phones analog phones connected on this side and then connecting over IP network and then we have actually twin redundant indexing servers we call it SIP proxies and then people are going to have their own wife phones they might actually put a conventional phone with a media gateway they can actually use a laptop or a machine a soft phone running on that all three possibilities will exist and depending on the capability of devices you can set up even video calls you can set up conference calls you can set up chat sessions actually email ideas technically can be used for doing all communication. So, when you want to make a phone call to an email ID. So, it will go to again the indexing server will find out for this email ID what is the corresponding number and then this will be controlling is in our case it is not controlling it is a SIP based system. So, it also is just visible as a 6000 wife phones connected through one single port, but if the PSTN switch would have been there if there would have been a exchanges network. So, you require a separate signaling between this. So, you require media gateway control in this case, because there is something more additional thing which are required that life is over right exchange will no more be there. We are just connecting existing 6000 5000 lines on to this media gateway the problem with this media gateway is again they are not coming in very large sizes they are mostly 24 port which are there. So, we have we will have lot of them stuck. So, 3000 will be here and 2000 will be there in hall 1 and they will be connected through IP network earlier we were actually having a separate fiber connecting to exchange. Over the time currently not because the people still have an issue if the power fails. See problem at a house if I put a wife phone I do not I will not be having ADSL. ADSL is being thrown out now. So, we have ethernet at every house and it is connected through optical fiber optical fiber cannot carry power we still do not have I think any mature technology by which I can also transfer power through optical fiber. I think we need to even have that kind of stuff. So, where actually we transmit signal as well as power through optical fiber optically and then we tap it out and use it for the switches on the other end. So, unless the fiber cut actually happens the power can still be transported currently the power is if transport is required you require either a separate power cable which has not been done in our case. So, it is a locally powered stuff, but with some kind of battery backup, but suppose a power fails in campus for more than 8 hours then there is a problem UPS will also age. So, some sites there will be disconnection because of that. So, those wife phones at houses will not work. So, you cannot even make a complaint to the IWT that my power is not coming. So, this is a very sensitive issue, this we have gone through ultimately we figured out we cannot we have to actually provide powering mechanism from the exchange side and that is why this media gateway has come into picture with conventional phones. So, every house in the campus will actually have two phones and technically it will have the same number and this will be an example of what I will talk about in SIP when a phone will come both the phones will ring and you can pick up any one of them and they are not parallel phones they are technically two separate phones and indexing server will now be sending message one to the gateway and one to the actual wife phone which are associated with the same number. See interesting thing is with the same IP address and port will have one possibilities for the same user ID I can register multiple devices. So, when number will come I this can be routed to all of them parallely can be probed any one of them is picked it is done this is known as forking in this case of SIP or one by one if you try this thing this does not respond within a time go to the next one go to the next one ultimately none of them are been picked up go to the voice mail box which is again maintained as a separate server and SIP will be routing the call to that is one possibility all of them can be done in parallely we will be actually using that now there is another interesting which can be done you are going to your friend's house you can go and actually type in a number and register even your phone number on that port now both the numbers are on the same port is like dual SIM system dual SIM phone. So, multiple identities can be there on single port that is possible ordinary phone ordinary analog phone which is already there as of now the big gateway we are already purchasing 6000 line that is already there that there are multiple of 24 ports actually they will be put in the rack there is issue of reliability and power see when I am putting everything in the exchange I have my good power backup source if I do it distributedly at 20 sites their hearts for example which have been built they have got a small ups now maintaining the number of the ups if they are more the more will be problem for you for maintenance I think the only problem with this system is this is not for the countries where power situation is bad that is the only thing which I feel is a problem I cannot transfer power I can I think over ADSL also it can be done, but usually ADSL ports are powered up locally all ADSL modems they are not powered through the telephone line, but if you can do that if you can actually make very low power ADSL modems which can take up power only from the telephone line itself then actually this is possible you should also now appreciate one more thing we call it loop unbundling I can change my operator I got a wire laid from telephone exchange of BSNL telephone exchange no more required you are going to for voice over IP only ADSL so that media is copper media is only used for data transport so voice is transported over ADSL over IP now so if at that exchange itself it is there I can actually now simply change my SIP provider my phone is still the same my port physical port is same IP address is same port number is same I can keep on changing my service provider at any point of time and I only pay to BSNL only for the loop and data transport charges and remaining charges I pay to the SIP service provider now here also issue of how you will do the billing will also come into picture currently what you do you get a telephone for every telephone you make the call make the payment 1 rupee per minute standard call rates now what you will do you are actually going to have a line you will pay for how many bytes which have been transferred and you will are you going to pay additionally to the SIP service provider that 1 rupee per minute his role is only when the call is set up once the call is set up you are directly talking peer to peer you no more require any more services so whether you talk for 1 hour whether you talk for 2 hours does not matter so it is how many times call will be set up per call set up the charges will be there by the SIP service provider and for the duration it means call will be free no it is not free if you talk for 2 hours you are transporting more bytes and you pay on per byte basis you are still paying to BSNL in that case but you can keep on changing your SIP service provider the way you want the way Vonage gateway people are actually using and it is a US number they are using SIP service provider of US it is I think fixed annual fee which is charged by in US actually by the companies but here you pay to BSNL for your data transport charge. ADSL is only a media link layer mechanism which provides a data transport it can when you are using Vonage it is going actually as voice over IP conventionally it is not because what happens from your phone you have an ADSL modem you have your machine or whatever your IP phone or Wi-Fi whatever it is this going over a single line you have it what we call a filter here there is a filter even before this modem actually there is a filter here so this is the way it goes actually so this is a in this direction it is band pass only lower band goes for this side it is a higher band which is not used by voice that will be used for data transport high pass filter this is a low pass filter on this side high pass on this side. So similarly there is a going to be a filtering so all high pass stuff will come low pass will go and this will be going for the exchange part where it is a line card so your voice can be routed through this conventional phones as far as this phone is concerned it does not see actually your digital part it does not see the ADSL thing it is just from this point onward to this point you are using common media but technically they are two separate networks physically they may be one because this line is maintained by the same guy and on top of it you are using voice over IP you are using your cable TV what is your IP TV we call it you are using your data connection all three so data voice and TV but I call it now actually multimedia so data voice TV everything will we actually first up and it is all kind of will mix into each other over time currently that is the way the BSN actually sells other stuff all three together triple play they call it now so triple play is for these three things. So you have to buy a separate set of box IP set of box here for connecting but I do not know whether it is the same ADSL modem plus set of box is common or it is separate I have never seen this set then logically it must be separate because it is an optional thing IP TV stuff so these four protocols will be actually used this simply will give a textual description of that for example I am sending a SDP message to you it will be sent through SIP mechanism and once I send a message I will say I want to actually have a phone call and I am going to use this part of the codec rate this is my capability because I am going to transport video I will they will not be any sorry they will be I am transporting audio no video I do not need a video call my bit rate maximum is this and I have these codecs available with me for example this message goes to you you will read this message and you have your own capabilities you will say yes I can make a audio call I have all that hardware with me you will confirm back and you will then find out what all audio codecs actually I have written what is my bit rate you know your capability whatever is common that common thing which what you are going to send it back to me. So common subset of the two will be sent back to me again in another SDP reply reverse also SDP comes and based on that I know we both are using the same rule. So this is the codec which I have to use I will trigger that codec you will also tell me I have to connect to your IP address is this your port number is this on which I have to connect for my media and once your thing come I will do acknowledgement and then you will set up a media stream and you will start talking and then we will disengage these guys the SIP service provider will not be aware of what I am doing actually but if he is routing the call in this fashion if there is intermediary gateway or then only he will know it media gateway also he might know it actually RTP is for signaling in real time for example I want to switch over RTP is again end to end it is nothing to do with SIP once I know my peers address I have a signaling thing I can start a RTP session so I can do a signaling on that so we will switch over the audio codec from higher to lower bandwidth depending on if my bandwidths are changing. So it is basically real time you will keep on telling me what is the performance of the reception whether something has to be dynamically changed streaming is I will be just maintaining what is called buffer control for example we do not use in our Braspati sync we are not using RTSP of course it is a bad design but that is the best which I could have done with whatever is existing manpower which I have in the project so we use buffer management if buffer over we actually crosses a certain flow we start dropping packets every third packet will be dropped if there again too much happen every alternate packet will be dropped or every tenth packet we start with ten actually every tenth packet will be dropped. So we keep the buffer within bounds if buffer actually empties out it means we are consuming at faster rate and buffer is incoming rate is lower so I have to now wait for some time so it is not that when the first packet comes I will start playing immediately I will wait I will wait for ten packets to accumulate and then I will start my player and audio and video need to be synchronized remember audio and video need to be synchronized time stamping is important there so when I am writing here something and I am speaking so all actions have to be synchronized so that is why what actually this streaming protocol actually does this is what is that feedback comes through RTP RTSP does not bother about feedback it does a synchronization part how many packets were dropped every packet will be numbered see the hundred packets were transmitted I got only these many these many media packets were dropped so you will try to make a guess based on that you remember you are doing UDP transport it is not TCP there is no retransmission even when you are doing signaling has to be done in real time can you do TCP you cannot you have to still do UDP UDP something is missed out crucial or not crucial that has to be seen well if we have actually for example are suffering from a situation we are actually currently in braspathism not using any one of these well this is not required there because it is a proprietary system these two could have been used this could have been used this we are not using we are using our own XML schema description proprietary this we are not using we are using buffer control here we do not bother if slight voice mismatch something happens you will always find the lip sync is a problem in braspathism Skype lip sync usually you would not actually figure it out they use the again their proprietary codec equivalent of which actually does provide synchronization of media streams synchronization I think is one of the major issues so it is always done through time stamping and creating separate buffers for each media stream and dropping and adding depending on the requirement real time I think what it means is another connotation of this is the clocks which I am going to use everybody's clock if you have actually done a GPS sync may be giving a GPS clock so everybody's clock will be uniform so today I have matched my clock with yours is 940 and tomorrow if I come at the same time your clock says 935 mine says 940 so 5 minute gap has been created this takes care of even that scenario when playback rate and recording rates are different they are bound to be different because all clocks are not synchronized over internet this sometimes what you call buffer overflow buffer underflow both problems can actually happen and but amazingly this is not perceived by us because when you play MP3 music for example on every computer the clock rate is different you buy 2 gigahertz but actually it is not 2 gigahertz 2 plus minus something which is there when you are playing the music you may not be actually listening to the original music is always slightly disturbed unless the clock rates are perfectly synced but humans should not be able to perceive only that much tolerance is there Zitter is going to be there you cannot remove that so far you can communicate it is fine okay now there is one more issue I would like to explain that and then come to the sip sometimes the problem of behind a proxy of behind a net that actually comes into picture in IIT Kanpur Skype does work actually without any issues even if you have behind a proxy so Skype if you go to tools options settings whatever it is and you will find there is a proxy setting there proxy username and password is also there okay so it is a STTP based transport what is STTP based transport STTP is a very unique kind of protocol it is I send a request you give me a response you forget about me after that so it does not remember what was my earlier request and what is my state so is a state less protocol there are no states maintained okay web servers of course do it actually maintain it through a different mechanism these are known as cookies session cookies basically through that which are sent with every request so it can figure out I am talking about which particular state information actually is transacted with every transaction so then it can remember that state on that basis using those pointers so it is a state less protocol so important thing that you want to send something to me how what happens in IITK suppose this is IITK network this is your machine this is a proxy this is the outside world I cannot send any information to you this is not a web server I am using private IP address so this guy cannot connect here so if I connect a Skype running here this Skype cannot send audio packets or video packets on to this side it is not possible it is always this is going to start the connection request the proxy for getting some information it will go to server and then information will come back this guy cannot initiate this stuff now that is a bottleneck here so far I have been using symmetric system both sides can independently push the information it will flow and to reach in this case they cannot reach it is not possible so what you will do is we have actually face this problem so we have used a technique here we used what we call time out mechanism when the periodic fetch so whenever you might have seen in your certain web pages when you download after some time it times out and it again refreshes the page there is a refresh interval so usually what happens whenever the STTP request is sent then the STTP response is sent in STTP response I actually can set a refresh interval so this guy is going to set the refresh interval and periodically keep on fetching now there again two methods in STTP one is get another one is post get method is you will send only the URL there is nothing is there in the payload part so you only can send everything in the URL itself and then only retrieve the information in response to the get and there is a post method in the post method I will have a URL I will also can add the payload part and there is a response to that post method I will get something in the payload so what actually usually Skype does or for that matter Braspati sync does client is periodically doing post method to a URL which responds to this and whatever packets which need to be transported on this direction will be sent as the payload of the post and when the response of that will be coming whatever were the packets which are queued up here will be sent in this now it does not if it is stopped sending post method that STTP request all queued up packets will remain queued up here they cannot be retrieved this is no way can send the packets only this has to push the packets and pull from there push the packets and pull from there this has to happen periodically at faster rate now that is the problem with STTP and STTP has its own overhead it has to set up a TCP connection all the way here fortunately for every request we do not now set up a TCP connection again and again usually you set up one single connection keep on sending a request on top of it and this will again set up a TCP connection all the way here and then STTP will be separate text based messaging on this TCP system so this is something which sometimes is required if you are behind a proxy and nothing else can be done even if you are behind a netting router then also it is not possible that this guy can send the information back here unless a static net entry gets created on this netting router so in IIT Kanpur it is not possible that somebody from outside can come in to and send information to ourselves we have to keep on pulling it out so your Skype which you are running in your machine is periodically pulling from some other Skype client which is running outside because your friend with whom you are talking might also be there behind a proxy this poor guy also can only push and periodically fetch these two guys when they need to communicate they require something outside without this this system cannot work it is encryption it is purely encryption I am going to come to how the authentication will be done in this case because I think Skype that is my guess I have not read the documents but the way it is operating I could actually figure out most likely this could be the one possible reason and I think the design should work so that is what I am also planning for our new Braspati 4 system as well as for Braspati sync clients ultimately we will actually move to that we will not be using no more login passwords login password will be used to get initial certificates only so Skype for example everybody starts doing login when the machine boots up is going to be a big problem on the server so I think they are using security certificates when you do first time login it takes some time once a certificates comes and gets installed everybody has authenticates by presentation of those certificates so you can authenticate with every your super peer with that you need not go to the central server so even if central server is down for some time services will not be off people we can still authenticate each other that is a good thing about that system it is actually server failure proof system even if server fails it is okay I will I will come to that now this is what is known as reflector reflector in we call it VR vs this is again a multimedia system which was built I think by lot of open source guys this we also cause call this same thing as a reflector in Braspati sync system this is known as super peer or super node in case of Skype and this actually means you are running a Skype you are connecting through your ADSL and you are using a public IP that technically means if you are not using HTTP proxy you are behind a netting router also it is fine because is usually the static net or sure in a public IP you run a Skype you are not making any call but still data is being transacted and remember your operator might be charging on per byte basis you are going to still pay without making any call so I think we close here and I will now come to the security architecture in the next class and then see after that