INFORMATION MANAGEMENT
COST EFFECTIVE COMMUNICATION WITH ENGINEERING SERVICE TEAMS WORKING ABROAD BASED ON CLOUD
COMPUTING AND SIP (SESSION INITIATION PROTOCOL)
Introduction
Members of engineering service teams and technical staff working abroad, in an international environment – which supervise, maintain or provide technical service for distributed manufacturing systems located in different locations (sometimes divided between two or more countries or distributed worldwide) need to have a cost effective, reliable and flexible voice communication system, compatible with existing telecommunication systems. Due to the ongoing need to communicate with one’s own company, as well as with external cooperating resources (eg. technical support, spare parts delivery, documentation services, maintenance support) – their voice telecommunication system must be compatible with ‘normal’ (stationary and mobile) phone systems, like:
– PSTN – Public Switched Telephone Network, – ISDN – Integrated Services Digital Network, – SS7 – Signaling System 7,
– 3G (3rd Generation) mobile networks,
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which are still in common use by the external cooperating companies listed above, suppliers and other support resources. The use of aforementioned technologies is still quite expensive due to the high rates for international calls and additionally for roaming costs while abroad.
On the other hand, modern manufacturing systems are usually quite well networked and most of them allow the possibility to connect IP-enabled devices to the local network with the ability to make WAN connections. This permits the use of open VOIP protocols; such as:
– H.323 Protocol,
– MGCP – Media Gateway Control Protocol, – SCTP – Stream Control Transmission Protocol, – SIP – Session Initiation Protocol,
– IAX – Inter Asterix Exchange Protocol,
for cost effective voice communication with engineering service teams while abroad.
This paper presents experiences of the practical use of Cloud Computing: SIP protocol, SIP-Service Provider and SIP-enabled devices for low cost, high quality international voice communication, compatible with standard telephone systems. The solutions presented here have been tested on a sample international route between two locations: Hamburg, Germany and Szczecin, Poland.
The main advantage of this SIP solution is the low cost, which allows flexible communication management, reliable voice communication and the decreased communication costs improve a company’s monthly expenses.
Session Initiation Protocol
Session Initiation Protocol (SIP)1 is a simple, text based control/signaling protocol defined by Internet Engineering Task Force2 in RFC 2543 and RFC 3261. SIP uses requests and responses to establish and to control communication (multimedia sessions) between SIP-users using SIP-Servers and SIP User Agents. SIP protocol runs on top of standard TCP, UDP or SCTP
1
J. Cumming, SIP Market Overview – An analysis od SIP technology and the state of the SIP Market, Dataconnection, http://www.dataconnection.com, 19.7.2009.
2
IETF Internet Engineering Task Force; RFC 2543 (1999) and RCF 3261 (2002); SIP: Session Initiation Protocol.
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transport protocols, and uses the regular services provided by other Internet application-layer protocols (DNS, LDAP, RTP and others).
Session Initiation Protocol (SIP) supports the following facets:3
– SIP user location (determining location of the end-system - to be used for SIP communication),
– SIP user capabilities (determining the media and media parameters to be used for SIP communication),
– SIP user availability (determining the willingness of the called party to engage in communications),
– SIP call set-up: “ringing” and establishing call parameters (at both: called and calling party),
– SIP call handling (including transfer and termination of calls).
SIP Registrars/SIP Service providers usually provide a complete service package, which realizes aforementioned typical SIP functions, but can also provide add-on functionality, such as:
– local phone numbers for phone communication,
– free-of-cost SIP-connections to other SIP-users within the same provider (in fact: free phone connections to phone numbers within same SIP-network), – possibility to make calls to all “normal” landline/ mobile phone numbers:
domestic and international – at competitive rates,
– possibility to receive phone calls from “normal” landline/ mobile phone numbers.
To utilize outlined SIP facets for communication between engineering service teams, the following components must be set up to work together: – own SIP-server (like: Asterix4
) – or alternate, based on Cloud Computing concept: SIP-Registrar/ SIP-Service Provider,
– SIP enabled devices (hardware terminals or software terminal emulators), – good quality access to global network through local networks at locations,
where engineering service teams are working/ provide their service.
There are a number of SIP-Service providers available in each country; however the final selection of a local SIP registrar is an important element of the presented implementation, due to:
3
CISCO, Overview of SIP, http://www.cisco.com, 19.7.2009.
4
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– technical aspects (the route from end-users to physical location of SIP-Server should be as short as possible),
– economic aspects: prices for outgoing external calls (domestic and international) should be as low as possible.
There are also a large number of SIP devices available on the market. Presently, SIP users can choose between numbers of:
– hardware solutions: SIP phones, SIP adapters (to connect “normal” phones), SIP Gateways, SIP-Routers,
– software solutions (Soft Phones) like: X-Lite,5
Kphone,6 Linphone,7 SJPhone8 and many other solutions.
Preferred devices are recommended by the local SIP service providers, which are already tested and approved for work with a particular SIP Server and SIP Provider.
Evaluation of key factors for SIP communication
The most important factors for communication based on SIP protocol are: quality of the local and global network connections, the hardware and software used by SIP users and the quality of service offered by local SIP Providers.
The following aspects and technical parameters of local and global TCP/IP networks are most important for communication using SIP:
– LAN quality – especially throughputs in the local network: maximum download and maximum upload speed allowed in local network. Most ISP (Internet Service Providers) offer ADSL (Asymmetric Digital Subscriber Line) with upload speeds much lower than download speeds. SIP communication is bi-directional and average symmetric, therefore lower speed limits should be considered (in such ADSL lines: lower limit = upload speed limit) when planning to establish the SIP connections through these ADSL lines,
5
ConterPath Solutions, X–lite Softphone Software http://www.counterpath.com, 19.7.2009..
6
KPhone – SIP User Agent for Linux, http://sourceforge.net/projects/kphone, 19.7.2009.
7
LinPhone – Open Source SIP video–phone for Linux & Windows – http://www.linphone.org, 19.7.2009.
8
SJ Labs SJPhone for Windows, Linux, MAC and PDA: http://sjlabs.com/sjp.html, 19.7.2009.
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– WAN quality: especially global data flow speed (on route between selected SIP service provider and SIP users). Some SIP service providers offer links to external Internet services to measure download speed to SIP users. Most of these Internet Services offer software measurement of upload/ download speed between SIP-user and pre-defined checkpoints. Those check-points are usually test-servers, not located in the selected SIP provider network, and therefore the measurements give only average results,
– SIP-hardware and software – especially quality of built-in codecs used for voice code and decode,
– functionality and quality of service offered by SIP-Registrar/ SIP-Service provider site (SIP-Server).
The LAN and WAN quality and network performance can be acquired by SIP users through the measurement of the following technical parameters: – download speed at SIP user location,
– upload speed at SIP user location,
– RTT (Round Trip Time) – the packet latency time (minimum, maximum and average latency) while uploading/ downloading,
– Jitter (variability of the packet latency) on route between SIP user and SIP Service Provider,
– TCP packets lost during upload/download operations.
Sample implementation of SIP communication and test tools
To establish voice communication based on the SIP-Protocol for the sample international route, between two major locations in Poland and Germany:
– engineering team’s head office was located in Szczecin, and most cooperating partners were located all around Poland,
– engineering team moved from location to location in Europe, while the presented test was temporary located in Hamburg.
The following components have been used and tested: – SIP enabled devices,
– hardware solution: LINKSYS SPA21029
– phone adapter with router, – soft phone: X-lite,10
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– SIP Registrar/ SIP Service Provided: IPFON11
with SIP-server located in Poland,
– regular ICP/IP access to internet in both locations via local networks. To measure and evaluate broadband speed and network performance at SIP user locations the following tests were performed using internet services: – OOKLA’s speedtest12
– for general upload/ download speed measurements at both SIP user locations,
– OOKLA’s pingtest13
– for general latency time and jitter measurements at both SIP user locations,
– Numion14
internet connection speed testing site – for distributed surfspeed test,
– Visualpulse for measurement of packet latency time, jitter and packets lost to selected SIP- Service Provider.
Packet latency time, jitter and packets lost on route between SIP users and SIP Service Provider are the most important for communications based on SIP technology:
– TCP latency time, jitter and packets lost on route from Hamburg to the SIP- Server in Poland were monitored using Visualpulse – sample results: see figure 1,
– TCP latency time, jitter and packet lost on route from Szczecin to the SIP- Server in Poland were monitored using Visualpulse – sample results see figure 2,
10
CounterPath Solutions, X-lite Softphone Software, www.counterpath.com, 19.7.2009.
11
IPFON - SIP Service Provider, www.ipfon.pl, 19.7.2009.
12
OOKLA Speed Test Internet Service, www.speedtest.net, 19.7.2009.
13
OOKLA Ping Test Internet Service, www.pingtest.net, 19.7.2009.
14
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Fig. 1. Latency time on route from Hamburg to SIP-server in Poland (Visualpulse15)
Outlined results show good quality of network connections on route from both locations to SIP-Server, therefore the predicted quality of voice communication was good. It was also experimentally confirmed after that, via real voice connections.
Benefits and capabilities of using SIP solutions
The main advantages of using SIP based solutions presented in this article are as follows:
1. No need to pay roaming fees – while travelling abroad, there are no roaming fees for incoming voice calls for called party (called party is in this case the engineering service teams member, being ‘on duty’ abroad) – all calls are ‘virtually’ local calls – from/to local, domestic numbers.
15
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2. No need to pay for international calls – calling party can call engineering service teams, which are abroad and moving from location to location, to their SIP-terminals still using the same local (domestic) phone numbers; therefore there is no need to make international calls for calling party. Eliminating the need to make international calls also means no extra cost for international calls for the calling party.
3. Free of charge calls within one’s own SIP network – besides a small monthly fee to the SIP-provider (which can be used to pay for phone calls or fax calls outside of SIP-network), connections within one’s own SIP network – between SIP users logged into same sip network – are usually free of charge.
4. Compatibility with other, already existing phone systems – phone calls outside of the SIP-network, to any regular phone number worldwide are possible and price plans offered by the SIP Service Providers for using SIP-to-PTSN gateways are reasonable and usually much lower than those offered by regular or mobile telecommunication companies. 5. Easy adaptation to local networks - SIP terminals are easily or
self-reconfigurable: after initial set-up, they can be easily moved to any other location without the need to reconfigure (only basic TCP/IP parameters need reconfiguration, and these may be realized automatically, using built-in DHCP (Dynamic Host Configuration Protocol).
6. Flexible and scalable – SIP systems are scalable and – depending on number of users and their individual communication needs – can be easily reconfigured or resized.
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Fig. 2. Latency time on route from Szczecin to SIP-server in Poland
Conclusions
Acquired results show that voice communication between users located in Poland and Germany may be achieved inexpensively using SIP-protocol and SIP-enabled devices. Average observed latency time did not exceed 34 milliseconds with small latency fluctuations and testing did not disturb the voice connections. The quality of voice communication was good. The implementation of Cloud Computing and SIP technology permitted fluent voice communication with engineering service teams working abroad and dramatically reduced voice communication costs.
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EFEKTYWNA KOMUNIKACJA Z INŻYNIERSKIMI ZESPOŁAMI