A Comparison of SIP with IAX an Efficient new IP Telephony Protocol

Transcription

1 A Comparison of SIP with IAX an Efficient new IP Telephony Protocol Malik Salahuddin Nasir, Kamran Saeed Dep. of Electrical Engineering, University of Engineering and Technology, Lahore, Pakistan {Engr.malik7 & Sabir Hussain, Muhammad Usama, Muabshir Saeed CS&IT dep. Superior University and Dep. of Electrical Engineering, UET, Lahore, Pakistan com Abstract As the communication technology has moved towards the merging of voice networks into data networks. This merging has become possible with the concept of VOIP(Voice Over Internet Protocol) which largely depends on some protocols like SIP(Session Initiation Protocol), H.323 and IAX(Inter Asterisk Exchange). SIP(Session Initiation Protocol) is the famous of all IP telephony protocols and is currently in use as a backbone for VOIP(Voice Over Internet Protocol) enabled Communication Networks. This paper emphasizes on the new features and working of IAX(Inter Asterisk Exchange) with single UDP stream. These features includes low overhead of IAX(Inter Asterisk Exchange) packets and static transmission port number. There is also a comparison of SIP(Session Initiation Protocol) with IAX(Inter Asterisk Exchange) which concludes few advantages of IAX(Inter Asterisk Exchange) over SIP(Session Initiation Protocol). At the end there is a suggestion of how to use IAX(Inter Asterisk Exchange) in network which is already working with SIP(Session Initiation Protocol) and to use both of these protocols together in the same network. Keywords Voice over IP, SIP, UDP stream, Port number, IAX I. INTRODUCTION Few years back there were two separate networks, one for data communication like accessing internet, sharing data, downloading files, surfing website and ing. While the 2nd one was voice network purely for voice communication providing dedicated link among users. The difference between these networks is the switching technique used. Packet switching is used in data networks while circuit switching is used for voice [1]. Packet switching works with IP in which an IP address is compulsory to connect to network and to access the services on internet. In circuit switching a path is dedicated between two users when connection is established among them [2]. Service providers were providing both data and voice services on a single wire to its users, but in the core of service providers voice and data services were treated separately. Mobile operators for GSM system also have separate core architecture for voice and data [3]. It was much costly and difficult to maintain two different networks within an organization. Merging of these two separate networks have removed these difficulties and shrink the size of network architecture from large number of devices to few devices. This merging of technology has become possible with the concepts of "Voice over IP" and "IP Telephony" [4]. IP based voice communication is made possible due to protocols like H.323 and SIP. H.323 was the first protocol to introduce IP telephony services within present ISDN networks. But the algorithm of this protocol was not too mature and results in raising few questions about its efficiency and reliability in large networks facilitating millions and billions of users at a time [5]. The deficiencies in the code of H.323 rises the motivation to work in the development of an efficient protocol and as a result of it SIP was introduced. SIP was the better IP telephony protocol for signaling and controlling multimedia communication sessions in data networks. SIP was capable of providing video calling facility along with voice [6]. There was huge industry acceptance that SIP will be the standard IP signaling mechanism for voice and multimedia services. SIP was tested under different network architectures and under different scenarios, because of its success it is now in use as a back bone IP telephony signaling protocol. As the demand of high speed communication and number of users increases SIP shows few of its limitations to fulfill the needs present era [7]. These limitations forces the researchers to modify the algorithm of SIP or to introduce a new one and this happens in the shape of IAX a new and efficient IP telephony protocol. The code of IAX is proved to be a bit better than SIP but it is still in testing phase [8]. There will be a detailed discussion about SIP and IAX further in this paper and we will point out some preferable features of IAX over SIP. II. SIP AND ITS WORKING SIP is an application layer protocol standardized by IETF and used for initiating multimedia sessions of voice, video, instant messaging and virtual reality. It is able to modify and terminate IP telephony calls or other media session over internet [9]. SIP works as a request and response protocol, dealing with requests from users and response from servers. In a VOIP enabled network SIP work as a signaling protocol to establish internet call between two end users, when call begins it act as a media control and monitoring protocol and when the call session ends it again act as a signaling protocol to terminate the call and to transfer the call time information to server [10]. SIP performs some other functions like User Location function to geographically locate the end users from different networks, Availability function to check the status of end users like keep alive messages etc, Capability function to ensure that the different end devices like PCs Mobile phones and IP phones are able to communicate with each other, Registration function to register new users with the servers and to discard the users from network [11] [12].

2 As SIP enables communication between PCs, Laptops, Mobile phones and IP phones present in different networks, it has to follow some mechanism for determining the correct path in data network through which a packet can flow from source to destination by minimizing the delay [13]. To fulfill this requirement SIP places the packet routing information into its headers [14]. It uses four types of header in its packets like: 1. Record Route Header 2. Route Header 3. Via Header 4. Contact header The packet format of SIP is given in Fig.1. It looks like the short version of IP packet [15]. This packet contains the address of RTP packet in its payload. SIP works with RTP protocol to transmit multimedia sessions [16]. It has two UDP stream one contains the signaling informationon along with the information of RTP packet and the second stream of UDP contains RTP session to transmit internet call. That is the reason SIP require multiple UDP ports to work, one UDP port for signaling and 100 alternative ports for multimedia transmission [17]. Due to multi port usage, it is impossible for SIP protocol to bypass Firewalls and even it is unable to use NAT feature [18]. some other messages in different scenarios like '182 Queued' is send by server to tell caller party that called party is not available or the link is down. '401 Unauthorized' shows that user is not registered properly with the server or user don't have authority to do specific task. '404 Not Found' this is a message from server to caller party that called number do not exist. '408 Request Timeout' a message by server to users that response time is passed now send a new request. '600 Busy Everywhere' this message is broadcasted by server when the medium is busy or there is a congestion on link. Fig. 2. SIP call flow [20] Fig. 1. Sip Packet Format [19] III. SIP CALL FLOW There are few messages exchanged between the clients and server to establish a call and to terminate it. The call flow between user A and user B is given in Fig..2. When user A wants to make a call to user B and don't know the IP of user B it sends an 'Invite' message to server. Server forward this invite message to user B and reply user A with '100 Trying' message. When user B receives the Invite message it starts ringing and give response to server by sending '180 Ringing' message. Server forward this message to user A. User B after picking the call send '200 OK' message to server and server transfer this message to user A. After forwarding '200 OK' message to user A server allocates the media between two users and RTP stream starts between them. At the same time user A sends 'ACK' message directly to user B. When the call ends 'END' message is send by one user to server and server transfer it to other user and in response user send '200 OK' message to clear the media and stop RTP stream. There are IV. IAX PROTOCOL IAX is an application layer protocol used for controlling and managing multimedia sessions over an IP network. It was created by open source community of Asterisk PBX(private branch exchange) and its primary target was to efficiently control only voice calls over internet, but it is also capable of holding video streams with it. Mark Spencer owner of Asterisk developed this protocol with the vision to decrease the complexity and to reduce the deficiencies of VOIP communication [21]. IAX is an "all in one" protocol because of its ability to transmit media and control sessions together within same protocol. Unlike SIP or H.323 it does not require the support of RTP protocol. It uses single UDP stream to transmit and receive both signaling and media over static internal port number '4569'. Using single UDP static port IAX can bypass easily through firewalls and no other protocols are required to enable NAT with it. There is no requirement of extra configurations in the core network using IAX protocol to pass Nat Firewalls. It uses less overheads than RTP protocol and requires less bandwidth. IAX uses only 20% overhead with 4 bytes over a packet while RTP uses 60% of overheads with 12 bytes on each voice packet. IAX also has an ability of multiplexing and tunneling multiple channels over a single link [22]. Data from multiple multimedia sessions are merged into single packet, thus reducing the IP overheads and reducing latency. By using G.729 compression codec multiple calls can be sent over single MB bandwidth, for example combining IAX with G.729 codec, 103 calls can be transmitted over 1 MB of bandwidth. This protocol supports native encryption using

3 AES-128 method. Unlike text commands in SIP, IAX uses binary data which is easily understandable by most of machines. All the signaling information is transferred only over data link layer. DTMF(duel tone mode frequency) tones are also send along with signaling information. Like SIP, IAX also has a mechanism of call flow. In IAX enabled communication, call creation and termination messages are exchanged directly among users, there is no involvement of server between them [23]. When a new user enters in a network, it gets register with the server and after that users are directed with each other in form of Peer-to-Peer connectivity. Let us take the example of call flow between user A and user B. When user A wants to communicate with user B, it dials its number and a 'New' message is send to user B along with DTMF tone. User B receives this message and respond with 'Accept' message. User A reply with an 'ACK'. When the device of user B starts ringing it sends 'Ring' message to user A and gets the 'ACK' in response from user A. After receiving the call user B tells the user A by 'Answer' message and in reply gets an 'ACK'. During conversation multimedia frames are exchanged among users. To terminate the call one user sends 'Hang-up' message to other user and after receiving 'ACK' session and line gets cleared. IAX complete call flow is shown in Fig.4. To check the peers during calls and after the call, IAX has the mechanism to exchange keep alive messages among the peers. During the call session one user sends a 'Ping' message to other user in order to check its availability and user which receives the ping message respond with 'Pong' message to ensure that, I am alive. When there is no call session running, peers send 'Poke' messages to their neighbors and respond with 'Pong' message. These messages also maintains the quality of service, by comparing the values of 'Jitter' and 'Dropped frames' enclosed in initial Ping/Poke frames with the Poke frames which are send in response [24]. A flow of call monitoring messages is shown in Fig. 3. 1) Fig. 3. Monitoring Messages [25] Fig. 4. IAX call flow [26] V. IAX MULTIPLE FRAME TYPE IAX works with single UDP stream to transmit both signaling information and data, but each stream has different frame structure, depending on the type of data sent and the sequence of packets. Different Frame formats used in IAX encoding are: A. Full Frame B. Mini Frame C. Meta Frame i. Meta Video frame ii. Meta Trunk frame iii. A. Full Frame: This type of Frames are send when a call or video session starts, because it contains signaling information along with data. These frames control initiation and termination of IAX video or audio calls. Header of these frames are of 12 bytes or 12 octets [27]. These frames are reliably send and require acknowledgement in reply. Header of Full Frame, which is shown is Fig.5 contains different fields such as: a) F bit: This field indicates that rather the frame is full or mini. If there is '1' under 'F bit' it means frame is full or if it is '0' then frame is mini. b) Source Call Number: This field is 15 bits long and contains the caller number in binary form. c) Time-stamp: Peers in call keep the record of call timing from the beginaing, when signaling informatio is trasnfered and till the end of call. This field is 32 bit long. d) R bit: Information about retransmission of frame is kept in this field. If it is '1' frame is retransmitted otherwise it is not.

4 e) Destination Call Number: This is also a 15 bit field having called party number in binary form. f) OSeqno: 8 bit field keeps the sequence of number of outbound streams. It is initially zero but keep on increasing as the frames are sent. g) ISeqno: Opposite to OSeqno, this 8 bit field contains the sequence number of inbound streams. Initially it also zero but keep on increasing as the frames are received. h) Frametype: 8 bit field contains the information about which type of message this particular frame carried. i) C bit: Keeps the record of remaining 7 bits next to it. C. Meta Video Frame: Clear from the name these frames are for video session transmission. This type of frames are similar to 'mini frames' with an additional field of 'Meta Indicator' in its header. This field covers 15 bits from total 6 byte of header. Meta indicator contains all '0' bit to distinguish meta frames from other type of frames. The 6 byte header format of Meta video frame is shown in Fig. 7. There is also a 'V' inside is header which differ meta video frame from meta trunk frame. For meta video frame indication it is set to 1. Fig. 7. Full Frame Header [29] Fig. 5. Full Frame Header [27] B. Mini Frame: This frame is similar to Full frame but it is shorter in size and contains only 4 bytes of header. Task of this type of frames is to just carry voice or video data in an already established call session. These frames are sent unreliably without any signaling information and do not require any acknowledgement in reply, because it carry real time transmission and cannot bother any dely. Mini frame header is shown in Fig. 6 with some fields similar to those in full frame such as: a) F bit: This field must contain '0' to indicate that frame is not full. b) Time-stamp: Peers in the call keep the record of call timing from the start of session started by this frame to the end of sesson. This field is 16 bit long. D. Meta Trunk Frame: The frame shown in Fig. 8 is meta trunk frame which is used for trunking multiple audio, video session from different peer on a single transmission link. IAX protocol uses this Meta trunk frame to transmit several multimedia sessions altogether using single header. This reduces the overheads and results in reducing the delays. Header format of this frame is an enhanced form of Meta video frame with 32 bit time-stamp field and two other fields, 'Meta Command' indicating rather frame is trunked or not, 'Command Data specifies the flags for options used with the trunked frame. Fig. 6. Mini Frame Header [28] Fig. 8. Full Frame Header [30]

5 VI. IAX VS SIP IAX has multi frame architecture, with different frames used under different scenarios, while SIP has fix frame architecture. IAX separates user authentication mechanism from Caller ID, but SIP performs only one of these mechanism [31]. IAX has the reliability factor because it contains clear separation of Layer 2 and Layer 3 working, due to this when a user ends a call the session quickly ends and the medium gets free, while using SIP it took some time to end the session and to clear the medium. IAX implementation is not much complex because it is encrypted using AES-128 and SIP is encrypted using ASCII code, which makes it bit complex. IAX uses single UDP stream to transmit both signaling and voice or video, on the other hand SIP uses one stream for signaling and multiple streams for voice or video transmission. IAX do not require any other protocol to work with it, but SIP uses RTP protocol for real time data transmission and RTCP protocol to control and mange this session [32]. Signaling and Calling in IAX can be done through server between two peers or can be directly done in Peer-to-Peer connectivity. Using SIP signaling is done thorough server and Calling is done directly between peers. IAX allows endpoints to check the validity of phone number. SIP does not support this feature. In IAX DTMF is send separately, while in SIP it is send with initial signaling information. Bandwidth usage in IAX is less because its messages are in binary form, while SIP messages are in text format. IAX uses one fix port 4569 to transmit both signaling and data but SIP uses one port for signaling and multiple ports for RTP transmission. Fix port number allows IAX to pass through Firewalls with simple configurations, but this is difficult in SIP. NAT problem doesn't occur with IAX due to single port usage. IAX uses G.729 audio compression codec which makes it to use the bandwidth more efficiently [33]. IAX is a specific purpose protocol can be used to send only audio, video and dial plan data. While SIP is a general purpose protocol any type of data can be send using it. IAX supports trunking and multiplexing to transmit data of different users together on same link. IAX is an open source VoIP protocol but SIP is an industrial standard protocol, standardized by IETF. IAX is only in use with open source ASTERISK PBX, while SIP is implemented by most of ISPs in the world [34]. CONCLUSION Above discussion clear the view that IAX is far much better than SIP. The main vision behind the development of IAX is to minimize the current problems faced in the field of IP telephony, but when SIP was developed vision behind it at that time was to deploy IP telephony. This is the major reason of these differences between these protocols.sip has now become the industrial standard in the world, that's why it is not possible to totally replace it with IAX. But to make VOIP communication reliable and efficient, one can use both of these protocols together working in same network. Either we can use SIP as leading signaling protocol between two autonomous system and IAX can be used for calling and signaling among end users or vice versa. But I recommend to use IAX among end users, because of its abilities of efficiently bandwidth utilization and less overheads used by it. By using IAX on access side we can minimize the delays, which is the biggest problem in IP calling. REFERENCES [1] [Online]. [2] (2015) [Online]. searchnetworking.techtarget.com/definition/circuitswitched [3] (2015) [Online]. series.htm [4] (2014) [Online]. ericcontent0900a. [5] (2014) [Online]. [6] (2015) [Online]. Resources Networking Glossary [7] (2014) [Online]. searchunifiedcommunications.techtarget.com/definitio n/ip-telephony

6 [8] (2013) [Online]. www2.garr.it/conf_05_slides/s_niccolini- IPtelephony.pdf [9] (2013) [Online]. [10] [Online]. [11] [Online]. [12] [Online]. [13] [Online]. [14] [Online]. [15] [Online]. [16] [Online]. [17] [Online]. orial/sip-tutorial?offer=briefcase#work [18] [Online]. /An-introduction-to-SIP-part-3 [19] [Online]. ml [20] [Online]. [21] [Online]. [22] [Online]. 8SECT-2.html [23] [Online]. [24] [Online]. [25] [Online]. [26] [Online]. [27] B.Capouch, E.Guy M.Spencer, "IAX: Inter asterisk exchange,", [28] Githhub. [Online]. [29] Digium. [Online]. lists.digium.com/pipermail/asteriskvideo/2007-june/ html [30] E.Guy, "IANA consideration for IAX protocol," Feb [31] [Online]. [32] [Online]. [33] [Online]. [34] [Online]. [35] [Online].