Page 2 - Transport and signaling protocol in VoIP network; Processing of sampled signal; Formats; RMOSI layer model; Name of Layer
Tab. 1 - Compression in VoIP 2.3 Transport and signaling protocol in VoIP network For telephony over IP based network we need not only transmit audio data also we need transmit a drive information about connection and formation and unformation of connection. The several voice data are transmitted mo...
Page 3 - Tab. 3 - Payload of Real Time Transport Protocol; indicates the version of RTP used
Firstly, VoIP doesn't use TCP because it is too heavy for real time applications, so instead a UDP (datagram) is used. Secondly, UDP has no control over the order in which packets arrive at the destination or how long it takes them to get there (datagram concept). Both of these are very important to...
Page 9 - Collector for RUDE
Hence we have to measure below stated parameters for test quality of IP networks for real- time depend services like as VoIP. QoS parameters are defined in recommendation ITU-T I.350 and ITU-T Y.1541. The parameters are: • Loss rate • Throughput • Delay • Distribution of delay • Distribution of dela...
Page 10 - Measuring with GNU utilities; Fig. 4 -Principle of measuring QoS with GNU based program
5.2.3 QoSplot Qosplot is a tool that takes as input a set of text log files created by CRUDE [9] and produces as output the data and command files for gnuplot [11], to plot diagrams depicting the QoS characteristics. 5.2.4 GNUPlot Gnuplot is a command-line driven interactive function plotting uti...
Page 11 - Autonmous celsium clock on all sides (very expensive); Solution of QoS test; electrical engineering in Zagreb. In Test i used:
5.3.1 The fidelity of Measuring The fidelity of measuring depend on synchronization of time between sides of measuring [12]. The CRUDE/RUDE can use synchronization of time from: • Autonmous celsium clock on all sides (very expensive) • GPS receiver (available) • LAN (simpler but less fidelity) We wi...
Page 12 - PCM; RUDE have a configuration file for example
5.4.1 Simulation of VoIP flows With simulation utilities like CRUDE/RUDE we can not hollow exactly simulate VoIP flows (with signalization etc.) but we can good approximate a real traffic. In Tab. 5 you can see the most applied codec for VoIP, bitrates and number of Samples per period. Codec Speed ...
Page 15 - Network
Recommendation ITU-T Y.1541 IP QoS Classes and Objectives QoS Classes and Objectives Network Performance Parameter Nature of Objective Class 0 Class 1 Class 2 Class 3 Class 4 Class 5 IP packet transfer delay (IPTD) Upper bound on the mean IPTD 100ms 400ms 100ms 400ms 1 s Unspec. IP packet transfer d...
Page 16 - Voice network with Asterisk PBX; Part of Exchange; Fig. 9 -Full Asterisk PBX exchange with VoIP, ISDN and POTS clients; Part of PBX exchange configuration and functions
6 Voice network with Asterisk PBX Now we will unreel a cheap, powerful and modular solution of local Voice network with Asterisk PBX described in chapter 4 on the figure 3. The Hw possibilities of Asterisk PBX are placed on the figure 9. The network we can portion to: • Part of Exchange • Part of V...
Page 17 - Hardware used for building of Asterisk; Machine; Tab. 9 - Number of concurrent calls on different machines
interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as ...
Page 18 - Tab. 10 -Power test of asterisk PBX on PII 400 MHz machine; UAC; Fig. 10 -Principles of measuring PBX loading
Power test PII 400, 256 MB RAM with average time of calls tcs=10 s, time of test ttst=120 ms, G.729 codec New calls during 1 sec. Concurent calls TotalCalls created Retranslate calls Successfull calls Failed calls Timeout calls Unaccepted calls Average response [ms] CPU [%] 1 11 121 0 121 0 0 0 10 1...
Page 19 - Master configuration file
6.1.2 Installation and configuration The source code is available from ftp.dignum.com as a tarball. After download and unpack the tarball we can run make to build it and then make install . After installation we have to optional configure the PBX. Configuration files are stored in standard directory...
Page 22 - , then no match is possible until the SIP client has
Where xxx is the username associated with the SIP client, or is an arbitrary name used by other configuration files to refer to this SIP device. Typically if a SIP phone has an extension number of 123, then its corresponding entry in this file will begin with [123]. Note that you still have to enabl...
Page 26 - Part of ISDN connections; Tab. 16 -Summary of ISDN cards supported by the kernel's ISDN driver
Phone Protocol Codec OS PBX functions iFone SIP/H.323 G.729A,G.723.1, GSM/AMR, G.722 Lin, Mac, Win No SJphone SIP/H.323 G.711au, GSM Lin, Mac, Win No X-Lite SIP G.711au, GSM, iLBC, SPX, Lin, (only under Wine), Win Yes KPhone SIP Not published rec. codec Lin. No LIPZ4 SIP Not published rec.Codec Lin....
Page 27 - Part of local interconnected ISDN phones; HFC-S PCI based cards; Tab. 17 -Cards do have the required chip set capable of NT-Mode; Scitel cards
6.4 Part of local interconnected ISDN phones At least two cards are required to interconnect internal telephones with external telephone lines. In order to connect telephones directly to an ISDN card, it must support NT-Mode. Then it is possible to use it as internal ISDN port. There currently there...
Page 30 - we probably want to define
6.5.1.2 Testing the card and the physical connection For testing of physical connection we can use Minicom or another program with similar functionality. After start of Minicom isdn0 (nowadays you have to use Minicom -s isdn0) we have to setting port (ctrl-AO), /dev/ttyI0 as the device. For modem we...
Page 31 - . In this file we fill the bind; Default context for incoming calls; Every SIP client have self account with specification of
; and the line is considered up. "Ring" means we wait until the ring cadence; occurs at least once. "Answer" means we wait until the other end picks up. ;;mode=answer mode=ring;mode=immediate ;; List all devices we can use. ;dtmfmode=asterisk ; Detect using Asterisk dtmfmode=aster...
Page 32 - Tab. 22 -Dial plan configuration for Sip clients in extension conf; For connection to ISDN network we have defined global variable
secret=xlite2callerid="2102 xlite2" <2102> host=dynamicnat=yes canreinvite=no disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw;allow=ulaw allow=alawallow=iLBC Tab. 21 -Configuration of SIP clients in sip.conf in PBX 6.5.3 Dial plan configura...
Page 35 - Appendix A; DLink; Air; DLink 11b Dlink 122 USB network adapter; or in an airport terminal while waiting to board your next flight.
Appendix A DLink Air Plus DWL900 AP+ The DLink Air Plus DWL900 AP+ is an enhanced 802.11b+ Wireless Network Access Point featuring advanced silicon chip design from Texas Instruments, utilizing their patented Digital Signal Processing te...
Page 36 - HP ProCurve Switch 2626 Switch
HP ProCurve Switch 2626 Switch Designed for mid-size enterprises that require a smaller, more cost effective switch without sacrificing performance, the Switch 2626 broadens the scope of HP's Ethernet switch offerings, providing a lower cost option as a part of the HP ProCurve Switch 2600 series. T...
