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Telephony Platform

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FreeSwitch: Open Soure SIP-based Telephony Platform

Vibha Kasturi
University Institute of Engineering and Technology
Panjab University
Chandigarh, India vibhakasturi@gmail.com Abstract— Next generation network is a network that delivers all telecommunication services (data, voice and video) using single packet based infrastructure. SoftSwitch, the kernel of next generation networks (NGN), a software that is used to bridge a traditional PSTN and VoIP by linking PSTN to IP networks and managing IP multimedia traffic. This paper gives an overview of SoftSwitch and summarize the currently deployed free and open source SoftSwitch i.e. FreeSWITCH in telecommunication industry. FreeSWITCH uses SIP protocol for communication between end terminals. This paper also gives an overview of SIP protocol and SIP proxy servers used by FreeSWITCH in end to end communication.

Keywords- Next Generation Networks(NGN) ,SoftSwitch, FreeSwitch,Session Initiation Protocol, SIP proxy server

Introduction

Next Generation Networks (NGN) is convergence of various networks such as the Public Switched Telephone network (PSTN), the packet data networks (Internet) and wireless communication networks, which can provide multimedia services including voice, data and video. NGN can be divided in three functional layers: Transport, Call Control & Signaling and Service & Application layer as shown in figure1. [1]
[pic]

Fig.1 Today’s Network and Next generation network SoftSwitch, the core element of NGN in Call Control & Signaling Layer, is built over a system composed of many devices distributed over IP network. IP SoftSwitch is a platform that allows implementing various types of Voice Over Internet Protocol (VOIP) services, with retaining shared, uniform management interface. The International SoftSwitch Consortium defines the term SoftSwitch as a “Media Gateway Controller” (MGC) which control network elements that mediate between, route or translate different data streams. To communicate with other SoftSwitch and with network end points, such as gateways, IP-phones, softphones, signaling protocols such as Base Station System Application Part (BSSAP), Mobile Application Part (MAP), H.323, and Session Initiation Protocol (SIP) are used.

The remainder of this paper is organized as follows: Section II presents architecture of SoftSwitch. Section III describes FreeSWITCH and section IV presents characteristics of FreeSWITCH. Section V describes the protocols used for FreeSWITCH. Conclusion is made in section VI.

Architecture of softswitch

SoftSwitch provides the call control functionality in IP network. A general architecture of SoftSwitch is shown in figure2.

SoftSwitch Consortium defines the following functions of SoftSwitch:

1 Call Control capability

Call control function with various signaling protocols, such as ISUP, H.323, Session Initiation Protocol (SIP) and Bearer Independent Call Control (BICC).

3 Media Gateway function

Media gateway function provides conversions between circuit-switched resources, such as lines and trunks and packet-switched network, such as IP and ATM.

5 Signaling Gateway Function

It provides conversion between SS7 signaling network and packet switched network

6 Application server function

Application Sever (AS) interfaces the call control function, media server function and signaling function. Application server function provides API’s for creating and deploying Services.

7 Media Server function

It provides access to specialized media resources, such as Interactive Voice Response (IVR), conferencing, text-to-speech and speech recognition.

SoftSwitch are available in market as open source as well as proprietary SoftSwitch, such as Asterisk, FreeSWITCH, Mysipswitch, and Sip Express Router (SER) are free and open source SoftSwitch. Many organizations implement their own proprietary SoftSwitch for Internet telephony that provide support for video and data for both fixed and mobile Networks.

freeswitch

FreeSWITCH is released as open source under MPL (Mozilla Public License) in 2006. It is a cross-platform telephony system designed to transfer multimedia data such as speech, video or text over the IP network. The initial design goals of FreeSWITCH were modularity, cross-platform support, scalability and stability. FreeSWITCH support various communication technologies such as Skype, SIP, H.323, Google talk etc. FreeSWITCH has built on following platforms • Linux(x86& x86-64) • Windows (MSVC 2008 & VC++ EE 2008) • Mac OS (Intel & OS) • Free BSD6

FreeSWITCH is a SIP based VoIP system generally deployed for call session management such as to create, modify and terminate multimedia sessions and Internet end points. The general architecture of SIP based VoIP deployed for FreeSWITCH is shown in Fig. 3. [3]

[pic]

characteristics of freeswitch

1 Distributed Architecture

FreeSWITCH is implemented in a distributed manner as opposed to traditional centralized manner of PSTN. In a distributed network processing takes place at various points throughout the network. Thus if a single point fails, the network operations remains unaffected. It also increase scalability

2 Threaded Model

FreeSWITCH has a threaded model to support parallel operations. Every connection has its own thread and events are pushed in to thread by event handlers.

3 Dynamic Modular Architecture

FreeSWITCH has a modular architecture. It installs with a basic set of modules for the usual iPBX (private branch exchange) tasks, and then additional features come from whatever modules user can find or code by own. Different modules can be used dynamically as:

• FSAPI interface allows the addition of commands that can be accessed remotely. • APPLICATION interface allows creation of custom IVRs and call routing apps. • XML interface allows real-time dynamic access for user directory and configuration. • Endpoint interface connects IM and phones. • Allows third party modules and extensions to customize application without having to start from scratch

4 Miscelaneous features

FreeSWITCH can act as media gateway, media server, and PBX or session border controller.

protocols used in freeswitch

FreeSWITCH use number of protocols such as Session Initiation Protocol (SIP), H.323, and IAX2. SIP is a standardized application layer protocol for session management in general e.g., for establishment of a VoIP call. It provides simple and efficient handling of multimedia sessions among multiple users. However, SIP itself is not a vertically integrated communication system, it is a part of overall IP telecommunication architecture which is composed of a range of packet switched protocols fig. 4. [4]

SIP is independent of underlying transport layer protocols; it can run on Transmission Control Protocol (TCP), User Datagram Protocol (UDP) or Stream Control Transmission Protocol (SCTP). The voice and video stream communication applications are carried over another application layer protocol, the Real-time Transport Protocol (RTP). SIP is a peer to peer protocol. The peers in a session are called user agents (UA). SIP specifies central network entities, SIP proxies which are responsible for routing SIP message. SIP proxy server accepts the session request sent by UA client and queries the SIP registrar server to obtain the recipient UA’s addressing and routing information [4]. SIP proxy server then forwards the session invitation directly to the recipient SIP UA server if it is in the same domain or to another proxy server if the UA resides in different domain. A SIP call setup when the user agents are in same domain is a shown in Fig. 5.
[pic]

SIP is based on an HTTP-like request/response model. A typical SIP message flow of a call session setup and teardown is shown in figure 6 [5]. To establish a multimedia session user agent client (UAC) sends an invitation (INVITE message) to proxy server to connect to the user agent server (UAS). Proxy server receives the UAS’s IP address and other routing information from SIP registrar server. If UAC’s invitation is accepted then UAS informs the SIP proxy server that it is ready to receive the packets. Through acknowledgement (ACK message) between UAC and UAS the session is established. After sending the packets the session is terminated by sending a BYE message either by UAC or UAS[pic] UDP is used to transfer SIP messages through configured UDP ports [6]. Each outgoing message is associated with a timer value, the expiration of which triggers a retransmission. UDP is also used to deliver audio packets during a voice call. When a SIP message is received, the parser module parses it and the values interpreted are passed to the UA server for an appropriate response. An outgoing message is composed by the message generator and transmitted using UDP.

In the current implementation of our work a call is generated to FreeSWITCH using Zoiper (Soft-phone). FreeSWITCH is act as a proxy server which forwards the calls from SIP user agent client to SIP user agent server. The current scenario is as shown in figure 7. SIPp is a performance testing tool for the SIP protocol. It includes a few basic user agent scenarios (UAC and UAS) and establishes and releases multiple calls with the INVITE and BYE methods.

Conclusion

The paper summarizes the information on SoftSwitch, a software counterpart of traditionally deployed circuit-switched networks. It also gives a brief introduction to free and open source SoftSwitch i.e. FreeSWITCH, its functions and SIP protocol implementation of FreeSWITCH for call session establishment between a user agent client and user agent server. A further study can be given on FreeSWITCH functionalities such as conferencing, IVR support, Management and support system, Billing and accounting system of FreeSWITCH

References

1] Wei Qiang, Su Sen, and Chen Junliang, “Study on application of SoftSwitch in wireless network” , Proceeding of ICCT2003 National natural science foundation of china, contact Number 60002006
2] S. Zeadally and F. Siddiqui, “Design and Implementation of a SIP based VoIP Architecture”, IEEE,2004
3] D. Malas, “SIP end to end performannce metrics”, IETF internet draft http://tools.ietf.org/html/draft-malas-performance-metrics-09, December 2007.
4] Zhang, Yuan, “SIP based VoIP network and internetworking with PSTN”, Electronics and Communication Engineering Journal, December, 2002
5] S. Zeadally, F. Siddiqui “Design and implementation of SIP based VoIP Architecture”, 18th International Conference on Advanced Information Networking and Application,IEEE,2004
6] Li huimin, liu Shufen, “A workflow based softswitching provisioning systen”,IEEE, 2009
7] Sureshkumar V. Subramanian, Rudra Dutta,”A study of performance and scalability metrics of a SIP proxy server- a practical approach”,Elsevier, 2010, Article in press.
8] http://www.Freeswitch.org
9] http://www.voip-info.org

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[pic]

[pic]

Fig. 2 Architecture of SoftSwitch

[pic]
Fig. 3 General architecture of SIP based VoIP system

[pic]
Fig. 5 SIP call within the same domain

[pic]
Fig. 6 SIP flow of message transaction

[pic]
Fig. 4 SIP-based IP telephony protocol stack

[pic]
Fig. 7 call generation using FreeSWITCH as proxy server

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