tag:blogger.com,1999:blog-77457109624660595872024-03-13T22:55:52.581-07:00Call Centerboonsrihttp://www.blogger.com/profile/03406139995152339005noreply@blogger.comBlogger4125tag:blogger.com,1999:blog-7745710962466059587.post-82525369864842083082007-12-15T08:44:00.000-08:002007-12-15T09:11:33.604-08:00<b>CAMEL</b> <p class="MsoNormal" style="margin-left: 18pt;"><o:p> </o:p></p> <p class="MsoNormal" style="margin-left: 18pt;"><o:p> </o:p></p> <p class="MsoNormal"><o:p> </o:p></p> <p style="background: white none repeat scroll 0% 50%; -moz-background-clip: -moz-initial; -moz-background-origin: -moz-initial; -moz-background-inline-policy: -moz-initial;"><span style="font-family:Arial;">CAMEL (Customized Application for the <st1:city st="on"><st1:place st="on">Mobile</st1:place></st1:city> network Enhanced Logic) is a standard for Intelligent Networks for mobile communications networks. It is currently deployed in all regions of the world, enabling mobile network operators to offer fast and efficient services to their subscribers. <o:p></o:p></span></p><p class="MsoNormal"><b><span style="font-family:Arial;">Standard Releases</span></b></p><p class="MsoNormal" style="margin-left: 18pt;">The ETSI GSM Releases<br />European standardisation activities</p><p class="MsoNormal" style="margin-left: 72pt; text-indent: -18pt;"><span style="font-family:Tahoma;"><span style="">–<span style=""> </span></span></span><!--[endif]-->CAMEL Phase 1 standardised by ETSI in 1996, as part of the GSM R96</p><p class="MsoNormal" style="margin-left: 72pt; text-indent: -18pt;"><!--[if !supportLists]--><span style="font-family:Tahoma;"><span style="">–<span style=""> </span></span></span><!--[endif]-->CAMEL Phase 2 standardised by ETSI in 1997, as part of the GSM R97</p><p class="MsoNormal" style="margin-left: 72pt; text-indent: -18pt;"><!--[if !supportLists]--><span style="font-family:Tahoma;"><span style="">–<span style=""> </span></span></span><!--[endif]-->In 1998, enhancements to CAMEL Phase 2 released by ETSI, as part of GSM R98</p><p class="MsoNormal" style="margin-left: 18pt;">The 3GPP 3G Core Network Releases…<br />Global standardisation activities</p><p class="MsoNormal" style="margin-left: 72pt; text-indent: -18pt;"><!--[if !supportLists]--><span style="font-family:Tahoma;"><span style="">–<span style=""> </span></span></span><!--[endif]-->CAMEL Phase 3 specified by 3GPP in 1999, as part of the 3G Core Network R99.</p><p class="MsoNormal" style="margin-left: 72pt; text-indent: -18pt;"><!--[if !supportLists]--><span style="font-family:Tahoma;"><span style="">–<span style=""> </span></span></span><!--[endif]-->3GPP Rel-4 does not contain new functionality for CAMEL.</p><p class="MsoNormal" style="margin-left: 72pt; text-indent: -18pt;"><!--[if !supportLists]--><span style="font-family:Tahoma;"><span style="">–<span style=""> </span></span></span><!--[endif]-->3GPP Rel-5 contains CAMEL Phase 4.</p><p class="MsoNormal" style="text-indent: 18pt;">CAMEL Application Part (CAP)</p> <p class="MsoNormal" style="margin-left: 18pt;"><span style=""> </span>CAP is a derivative of ETSI CS1</p> <p class="MsoNormal" style="margin-left: 72pt; text-indent: -18pt;"><!--[if !supportLists]--><span lang="FR" style="font-family:Tahoma;"><span style="">–<span style=""> </span></span></span><!--[endif]--><span style="" lang="FR">ITU-T CS; Q.1210 series (Core INAP CS1)</span></p> <p class="MsoNormal" style="margin-left: 72pt; text-indent: -18pt;"><!--[if !supportLists]--><span lang="FR" style="font-family:Tahoma;"><span style="">–<span style=""> </span></span></span><!--[endif]--><span style="" lang="FR">ETSI EN 300 374 series (ETSI CS1)<o:p></o:p></span></p> <p class="MsoNormal" style="margin-left: 72pt; text-indent: -18pt;"><!--[if !supportLists]--><span lang="FR" style="font-family:Tahoma;"><span style="">–<span style=""> </span></span></span><!--[endif]--><span style="" lang="FR">ETSI EN TS 101 046 - GSM TS 09.78 (CAMEL Application Part, CAP V1, V2)<o:p></o:p></span></p> <p class="MsoNormal" style="margin-left: 72pt; text-indent: -18pt;"><!--[if !supportLists]--><span lang="EN-GB" style="font-family:Tahoma;"><span style="">–<span style=""> </span></span></span><!--[endif]--><span style="" lang="EN-GB">3GPP TS 29.078 (CAMEL Application Part, C</span><span style="" lang="EN-GB">AP V3, V4)<o:p></o:p></span></p> <p class="MsoNormal" style="margin-left: 18pt;"><span style=""> </span>Camel Phase 1</p> <p class="MsoNormal" style="margin-left: 72pt; text-indent: -18pt;"><!--[if !supportLists]--><span style="font-family:Tahoma;"><span style="">–<span style=""> </span></span></span><!--[endif]-->Number Translation, Call redirection</p> <p class="MsoNormal" style="margin-left: 18pt;"><span style=""> </span>Camel Phase 2</p> <p class="MsoNormal" style="margin-left: 72pt; text-indent: -18pt;"><!--[if !supportLists]--><span style="font-family:Tahoma;"><span style="">–<span style=""> </span></span></span><!--[endif]-->Charging Control, User Interaction</p> <p class="MsoNormal" style="margin-left: 18pt;"><span style=""> </span>Camel Phase 3</p> <p class="MsoNormal" style="margin-left: 72pt; text-indent: -18pt;"><!--[if !supportLists]--><span style="font-family:Tahoma;"><span style="">–<span style=""> </span></span></span><!--[endif]-->GPRS Control, SMS-MO, Mobility Trigger, SS interaction, Dialed service</p> <p class="MsoNormal" style="margin-left: 18pt;"><span style=""> </span>Camel Phase 4</p> <p class="MsoNormal"><o:p> </o:p></p> <p class="MsoNormal" style="text-indent: 18pt;"><!--[if !supportLists]--><span style="font-family:Tahoma;"><span style=""> –<span style=""> </span></span></span><!--[endif]-->Mid-Call trigger, Call-party handling, SCP initiate, SMS-MT<br /><br /><b><span lang="EN-GB" style="font-family:Arial;">CAMEL Phase 1 Architecture</span></b><b><span style="font-family:Arial;"><o:p></o:p></span></b></p> <p class="MsoNormal" style="margin-left: 18pt;"><span style=""> </span>Basic control of <st1:state st="on"><st1:place st="on">MO-</st1:place></st1:state>, MT- and MF-calls</p><p class="MsoNormal" style="margin-left: 54pt;">-<span style=""> </span>No announcements</p> <p class="MsoNormal" style="margin-left: 54pt;">-<span style=""> </span>No prepaid charging</p> <p class="MsoNormal" style="margin-left: 18pt;"><span style=""> </span>AnyTimeInterrogation</p> <p class="MsoNormal" style="text-indent: 18pt;"><span style=""> </span><b><span style="font-family:Arial;"><o:p><br /></o:p>Network Overview</span></b></p><p class="MsoNormal" style="text-indent: 18pt;"><a onblur="try {parent.deselectBloggerImageGracefully();} catch(e) {}" href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEi7YH5jRe2-_oEI2S_q-H9UjftdQchyaRIGc79Tcx6LukpC0-weLfsOkueIFhyphenhyphenovCPmZLLdanT6CbEdvsxThbJTOqAMy1oKH0iM585S04kQxy-PfkNYQ6WfhwMtY1yDzOcZsXSL1aKT3EIp/s1600-h/camel0.JPG"><img style="margin: 0px auto 10px; display: block; text-align: center; cursor: pointer;" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEi7YH5jRe2-_oEI2S_q-H9UjftdQchyaRIGc79Tcx6LukpC0-weLfsOkueIFhyphenhyphenovCPmZLLdanT6CbEdvsxThbJTOqAMy1oKH0iM585S04kQxy-PfkNYQ6WfhwMtY1yDzOcZsXSL1aKT3EIp/s320/camel0.JPG" alt="" id="BLOGGER_PHOTO_ID_5144242792340786450" border="0" /></a></p><p class="MsoNormal" style="text-indent: 18pt;"><a onblur="try {parent.deselectBloggerImageGracefully();} catch(e) {}" href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEj6Jyzmim9jUmEbO4E_nXTIAjdgyOg2-38Ddum91MiaasqEDMHGy8zXDqy_70LlW_VnEBjSz9PlKkSPd4ksUpDKGLjZv1Z5Be97RX-Z8jlmTasps1vn8ttrxiUOgduVp5WRBQ2_8wqFVzo5/s1600-h/camel1.JPG"><img style="margin: 0px auto 10px; display: block; text-align: center; cursor: pointer;" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEj6Jyzmim9jUmEbO4E_nXTIAjdgyOg2-38Ddum91MiaasqEDMHGy8zXDqy_70LlW_VnEBjSz9PlKkSPd4ksUpDKGLjZv1Z5Be97RX-Z8jlmTasps1vn8ttrxiUOgduVp5WRBQ2_8wqFVzo5/s320/camel1.JPG" alt="" id="BLOGGER_PHOTO_ID_5144244351413914914" border="0" /></a></p> <p class="MsoNormal"><st1:place st="on"><st1:placename st="on"><b><br /></b></st1:placename></st1:place></p><p class="MsoNormal"><st1:place st="on"><st1:placename st="on"><b>Basic</b></st1:placename><b> <st1:placename st="on">Call</st1:placename> <st1:placetype st="on">State</st1:placetype></b></st1:place><b> Machine Principle</b></p><p class="MsoNormal" style="text-indent: 18pt;"><a onblur="try {parent.deselectBloggerImageGracefully();} catch(e) {}" href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEh8wAkINVt_wFwBbmUPKRIVh4YJBI1ZGG5j_Hg0b-BI86e9kdhvNzg-M0XN_Gpv5aVrRGkalA-YYjUdmUtL85S-nplQchxfuUf3OWGfprjs8Lg2p6tH7Nck135dafpHXETsqN5N-FCYTzDn/s1600-h/camel2.JPG"><img style="margin: 0px auto 10px; display: block; text-align: center; cursor: pointer;" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEh8wAkINVt_wFwBbmUPKRIVh4YJBI1ZGG5j_Hg0b-BI86e9kdhvNzg-M0XN_Gpv5aVrRGkalA-YYjUdmUtL85S-nplQchxfuUf3OWGfprjs8Lg2p6tH7Nck135dafpHXETsqN5N-FCYTzDn/s320/camel2.JPG" alt="" id="BLOGGER_PHOTO_ID_5144245270536916274" border="0" /></a></p> <p class="MsoNormal" style="margin-left: 18pt; text-indent: 18pt;">- Call handling is specified in GSM TS 03.18 ("Basic Call Handling"). Interaction between the BCSM and the gsmSSF is specified in GSM TS 03.78 ("CAMEL stage 2").</p> <p class="MsoNormal" style="margin-left: 18pt;"><span style=""> </span>- BCSMs have Detection Points (DPs) and Call States. The SCP may be contacted at DPs for Service Logic invocation or for instructions.</p> <p class="MsoNormal" style="margin-left: 18pt;"><span style=""> </span>- For every MO Call or MF Call, O-BCSM invoked in MSC or GMSC.</p> <p class="MsoNormal" style="margin-left: 18pt;"><span style=""> </span>- For every MT Call, T-BCSM invoked in GMSC.</p> <p class="MsoNormal" style="margin-left: 18pt;"><span style=""> </span>- CAMEL Phase 1, Phase 2, Phase 3 and Phase 4 have different BCSMs (more TDPs/EDPs).</p> <p class="MsoNormal" style="margin-left: 18pt;"><span style="" lang="NL"> - Three types of DPs exist:<o:p></o:p></span></p> <p class="MsoNormal" style="margin-left: 54pt;"><span style="" lang="NL">Trigger Detection Point (TDP)<o:p></o:p></span></p> <p class="MsoNormal" style="margin-left: 90pt;"><span style="" lang="NL">-<span style=""> </span>TDP may be statically armed (in O-CSI or T-CSI).<o:p></o:p></span></p> <p class="MsoNormal" style="margin-left: 90pt;"><span style="" lang="NL">-<span style=""> </span>CAMEL Service may be invoked from a TDP, provided that trigger conditions, if available, are fulfilled.<o:p></o:p></span></p> <p class="MsoNormal" style="margin-left: 54pt;"><span style="" lang="NL">Event Detection Point – Notify (EDP-N): <o:p></o:p></span></p> <p class="MsoNormal" style="margin-left: 90pt;"><span style="" lang="NL">-<span style=""> </span>EDP-N may be dynamically armed within a Service Logic.<o:p></o:p></span></p> <p class="MsoNormal" style="margin-left: 90pt;"><span style="" lang="NL">-<span style=""> </span>When an EDP-N is met, the SCP is notified. Call processing continues.<o:p></o:p></span></p> <p class="MsoNormal" style="margin-left: 54pt;"><span style="" lang="NL">Event Detection Point – Interrupt (EDP-R):<o:p></o:p></span></p> <p class="MsoNormal" style="margin-left: 90pt;"><span style="" lang="NL">-<span style=""> </span>EDP-R may be dynamically armed within a Service Logic.<o:p></o:p></span></p> <p class="MsoNormal" style="margin-left: 90pt;"><span style="" lang="NL">-<span style=""> </span>When an EDP-R is met, the SCP is notified. Call processing is suspended. The SSF waits for instructions from the SCP.</span></p> <p class="MsoNormal"><o:p> </o:p></p>boonsrihttp://www.blogger.com/profile/03406139995152339005noreply@blogger.comtag:blogger.com,1999:blog-7745710962466059587.post-31403366242903272532007-08-24T00:58:00.000-07:002007-10-27T12:08:44.367-07:00Next Generation Networks (NGN)<strong>Next Generation Networks (NGN)<br /></strong><br />Next Generation Networks are based on Internet technologies including Internet Protocol (IP). For voice applications one of the most important devices in NGN is a Softswitch (The Softswitch also acts as an MGC) a programmable device that controls Voice over IP (<a title="VoIP" href="http://en.wikipedia.org/wiki/VoIP">VoIP</a>) calls. It enables correct integration of different protocols within NGN. The most important function of the Softswitch is creating the interface to the existing telephone network, PSTN, through Signalling Gateways (SG) and Media Gateways (MG). However, the Softswitch as a term may be defined differently by the different equipment manufacturers and have somewhat different functions.<br /><br /><strong>Media gateway controller (MGC)<br /><br /></strong><strong></strong><img id="BLOGGER_PHOTO_ID_5102175074991397106" style="FLOAT: left; MARGIN: 0px 10px 10px 0px; WIDTH: 373px; CURSOR: hand; HEIGHT: 249px" height="234" alt="" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhL23wXC2u5zCJsFhSwBJoXNIW1TqO4ls9_5v5-BToFJt6EBKAtORinfQ9RB833A51MSwdtJtGbXmziDtOAOqNN1pBKMIKW-64-C8gXYoLomIbR2sSVSJpX4WRJQ762XgxuRt36F2LdN8My/s320/mgc.bmp" width="380" border="0" />MGC is residing on the control layer of Next Generation Network architecture, providing multi-service and packet voice connection control, routing, management of network resource, billing, authentication and all other functions. Secondly, MGC should interworking with other network elements such as trunking gateways, signaling gateway, multimedia application servers, other MGC and the existing/others SCP (for IN services) using standard protocols. MGC has to be able to support basic class 4 and class 5 functionality such as basic SPC services, some of advanced IP services, management & control of IP phone (both SIP and H.323) and Integrated Access Device (IAD) as well.<br />MGC has to support the following item.<br /><br />Interfaces:<br />The MGC shall support the following interfaces:<br />-IEEE 802.3x with 100 Base TX Ethernet for carriage of MGCP and ITU-T H.248 and SIP and SIP-T and H.323 signaling to core packet network.<br />-IEEE 802.3x with 100 Base TX Ethernet for communication to NMC and any OAM terminal.<br />-IEEE 802.3x with 100 Base TX Ethernet for Input/Output functions including download of billing data, operational measurements, traffic statistics, logs and alarm information.<br />-All 100 Base TX Ethernet inter must comply with IEEE 802.3x full duplex mode operation.<br />-E1 physical G.703 attached to STP exchanges in case of integrated functional of SS7 gateway.<br /><br />Redundancy and Reliability<br />-MGC must be a fault tolerant design. MGC should be highly reliable and highly stable. It should be built around a hardware platform delivering carrier grade, that is ‘five nines’ 99.999% availability.<br />-Essential components of MGC such as main processor board, communication board, power supply and so on must follow redundant design and be able to support insertion or removal with power on (hot swappable).<br />-The active to standby and vise versa switchover of main control part of MGC either causes from automatically by the MGC itself or manually switchover by the operator shall not affect any services and current calls and any call detail record (CDR) running in progress.<br />-MGC platform should adopt the dual-host system, work as hot backup mode. When the active host is faulty, the standby host will automatically switch over and take charge of the original active host’s function. And the switchover time should be less than 15 seconds to ensure the system reliability.<br />-All data storage devices should be mirrored disks or configured as RAID.<br />-MGC platform should adopt hot-backup mode based on highly reliable system with hardware and software fault tolerance design.<br /><br />Major Functionality<br />-The MGC shall have integrated function of SIP proxy server, SIP redirect server and SIP registrar in order to comply with IETF RFC 2543 to be able to manage and control the difference SIP endpoints, e.g. SIP user agent or SIP phone.<br />-The MGC shall be able to support the H.323 trunk side so the MGC shall be interoperable with others gatekeeper in H.323 network.<br />-The MGC shall be able to support managing and controlling of its own difference H.323 endpoints, e.g. H.323 phone and H.323 softphone by using H.323 and its protocol suites.<br />Protocol supported<br />-IETF SIP<br />-IETF SIP-T (SIP for Telephony)<br />-MGCP and H.248<br />-H.323 V.4<br />-PARLAY API V 3.0<br />-JAIN<br />-CORBA<br />-SIGTRAN<br />-SCCP/TCAP/INAP(CS1&CS2) in case of integrated SS7 signalling gateway to interconnect to others SCP.<br />-SNMP<br />-ITU-T Standard White book ISUP<br />-ITU-T ISUP (TOT’s version).<br />-The MGC shall be able to support PARLAY or JAIN (Java Application for Integrated Network) or CORBA (Common Object Request Broker Architecture) or SIP (Session Initiation Protocol) for Application Program Interface to inter-operate with the third party server who is providing variety of IP services/features.<br />-The MGC shall support Time of Day synchronization via an NTP server<br />-As a minimum the MGC must effect call set-up, control and tear down using MGCP (RFC 2705) or ITU-T H.248 signaling to:<br />Access Gateway<br />Trunk Gateway<br />Media Server<br />-If the present MGC – TG call setup protocol is IETF MGCP, bidder shall offer the option for upgrading overall system (MGCs, TGs and NMS) to ITU-T H.248 (MEGACO)<br />-The MGC shall also support ISDN signaling using IUA over the packet network to Trunk Gateways supporting connection of ISDN PRI services.<br />-MGC shall be able to effect call set-up and tear down between any of the following network elements:<br />Recorded announcement on Media Server.<br />ISUP Trunks supported on Trunk Gateway.<br />PRI Trunks supported on Trunk Gateway.<br />Tones or recorded announcements on Trunk Gateway<br />-The MGC shall be able to decode and generate all ISUP messages, perform any functions indicated by an ISUP message, and create any suitable response.<br />-The MGC shall support, and be compliant to, Blue Book ITU-T Recommendations E.164 addressing schemes.<br />-The MGC must support termination of call to tones or recorded voice announcement on trunk gateway according to the nature of the failure of the call.<br />-MGC must have a management of IAD and IP phone, SIP Phone, H.323 soft phone.<br />-MGC must have a management of IAD and IP phone, SIP Phone, H.323 soft phone.<br />-MGC software upgrading should be conducted online without any interruption of the running services, and online services loading should also be possible.<br />-MGC shall have traffic management function and self-adaptive overload control capability. According to overload cause and conditions, the corresponding algorithms should be dynamically used to carry out different degrees of filtration of service calls, to ensure safety of network operation.<br />-MGC shall control TG or Access gateway (via MGCP/H.248 protocol) even though the TG and Access gateway is another supplier’s.<br />-MGC must be able to support at least basic SPC functions such as :<br />Call Waiting<br />Caller Identification Presentation/Restriction<br />Caller ID Presentation on Call Waiting<br />Multiple numbering/identification on same device.<br />Call Conferencing<br />Selective Incoming Caller ID Restriction<br />Outgoing Call Barring<br />Call Forwarding<br />Close User Group<br />Anonymous Call Rejection<br />Carrier Pre-Selection<br />Abbreviate Dialing<br />Access to operator for assistance and information<br />Access to recorded announcements<br />Access to community services (public utilities and emergency calls)<br />Payphone service<br />Private branch exchange<br />Malicious call identification service<br />Interception service<br />Line observation service<br />Call barring service<br />Priority service<br />Hot line service<br />Do not disturb service<br />Automatic call repetition service<br />Subscriber with private meter service<br />Centrex service<br />Virtual private network (vpn) service<br />Absent subscriber service<br />Reminder (alarm) call service (call wake up)Last<br /><br /><strong>Trunk gateway (TG)</strong><br />Trunk gateway provides the direct translation to/from the TDM media of traditional (legacy) carrier infrastructures such as voice, fax DTMF, special tone, modem call to/from packet media.<br />TG has to support the following item.<br /><br />Major Functionalities<br />- TG shall be able to correctly compatible with the existing PSTN and VOIP network.<br />- TG shall be able to support completely remote software upgrading from centralize unit such as MGC.<br />- TG shall be able to comply with SDP Protocol (Session Description Protocol, RFC 2327) in order to communicate with MGC by using MGCP Protocol.<br />- TG shall be able to automatically detect all types of traffic such as: Voice, Fax DTMF and Modem call and must be able to serve these traffic by using any port.<br />- TG shall be able to support VoIP – modem relay function in order to carries modem traffic from ingress to egress gateways over IP network according to standard ITU-T V.150.0 and V.150.1<br />- TG shall be able to support voice activity detection (VAD) which will detect and suppress the silence in voice call of the caller at sender trunk gateway.<br />- TG shall be able to support Comfort Noise Generator (CNG) which is complementary function to VAD expressed. Its function is to insert some signal to called subscriber line to give them pleasant impression that the conversation still active even during the sender TG sending nothing.<br />- TG shall be support SIGTRAN (ITU-T Signalling System No.7 over IP),<br />- The TG shall support IUA (ISDN User Adaptation) to carry ISDN Dchannel signaling information from connecting E1 PRI to the MGC.<br />- TG shall support at least IETF MGCP and must be able to upgrade to fully support ITU-T H.248 signaling from the MGC to effect voice path connection across the core packet (IP) network.<br />- TG shall support at least the following CODECs (the conversion of TDM voice to packet):<br />ITU-T G.711 (u-law and A-law) Compression rate 64 kbps<br />ITU-T G.723.1 Compression rate 5.3and 6.3 kbps<br />ITU-T G.726 (ADPCM) Compression rate 16 , 24 , 32 , and 40 kbps<br />ITU-T G.729A (CS-ACELP) Compression rate 8 kbps<br />ITU-T G.729B (CS-ACELP) Compression rate 11.8 kbps<br />- TG shall support ability to pre-define type of CODEC respect to type of traffic or user administrator’s decision.<br />- TG shall support ITU-T G.168 echo cancellation with at least 128 milli-second tail.<br />- The TG shall support DTMF tone detection, generation, collection and transmission of the decoded information.<br />- The TG shall have the ability to modify packetization rate as 10 ms, 20 ms and 30 ms respect to the using CODEC.<br />- TG shall be able to deliver announcements/tones to connecting TDM trunks under instruction from the MGC.<br />- TG shall have ability to dynamic programmable of jitter buffer.<br />- TG shall be able to support both T.37 and T.38 Fax transport over IP network.<br />- TG shall support time synchronization using an NTP (Network Time Protocol) client. At boot and thereafter on a regular basis the TG should synchronize its date and time with a network NTP server.<br />- TG must support synchronization/clocking to an external timing source. At least three levels of synchronization must be supported with the final level being an internal clock source of Stratum 3 performance or better.<br />- TG shall be controlled by the MGC (via H.248 protocol) even though the MGC is another supplier’s.<br />- TG shall support feature DTMF delay according to RFC 2833.<br /><br />Traffic Management<br />- TG shall support the RSVP (ReSource Reservation Protocol – RFC 2205) architecture to establish path between endpoints (source and destination TG or IAD) together with core routers, with the goal of delivering high QoS on a media stream along the path.<br />- TG shall support the DiffServ (Differentiated Services - RFC 2475) architecture that source endpoint shall be able to specify TOS(Type of Services) bits in standard IP packet header to define classes of service that is method by which packet of a classification will be queued , forwarded and/or considered for discard during congestion when those packets reach DiffServ Node, core routers.<br />- TG shall support ability to manage the size of jitter buffer to preserve a good quality of packet voice at receiving endpoint.<br />- TG shall support IP/UDP/RTP/RTCP header compression function for reduce end-to-end packet overhead.<br /><br />Maintainability & Reliability<br />- The TG shall have the ability to accept the request for a graceful shutdown of the entire gateway, completing all active calls before the gateway is taken off-line for maintenance.<br />- The TG shall offer real time reporting of performance related information (i.e., performance threshold crossing alerts related to service affecting conditions on individual connections). The statistics to include the following and the information is to be sent to the MGC.<br /><br />Interface<br />- The TG shall support the following interfaces to narrow-band PSTN network for carriage of ITU-T Signaling System number 7, ISUP signaling,<br />- The TG shall support the E1 interfaces to a narrow band TDM network for the carriage of ISDN PRI trunks<br />- The electrical interfaces to PSTN side shall be complied with ITU-T G.703<br />- The TG shall support the multiple ethernet IEEE 802.3x (full duplex flow control with 100 Base-TX) to a communicate with IP core network for the carriage of converted IP packets traffic<br /><br />Redundancy<br />- shall support 1:1 redundancy on primary controller.<br />- TG shall support 1:1 redundancy on packet network interfaces, with automatic changeover in the event of the detection of a circuit failure.<br />- TG shall support 1:1 or N:1 redundancy on voice processing circuitsboonsrihttp://www.blogger.com/profile/03406139995152339005noreply@blogger.comtag:blogger.com,1999:blog-7745710962466059587.post-3254965901836476542007-07-23T07:35:00.000-07:002007-07-23T07:39:43.893-07:00The Intelligent Network Evolution<strong>The Intelligent Network Evolution</strong><br />Today, the main existing network of GSM Operator is Intelligent Network for prepaid service. They have distributed the Service Switching Point (SSP) to locate all the regions of country for serving GSM ‘s prepaid subscribers. By evolutionary extension of the service component's functionalities (and corresponding enhancement of the IN architecture) more advanced telecommunication services can be realized, while still supporting existing services. In this case, the existing IN shall integrate or inter connect with the new technology platform, new protocol and external rating engine for supported new services. And IN services are considered to include also mobility, broadband and multimedia applications. That is, the IN concept is considered to provide the necessary openness in order to enable the realization of future services. GSM operator look for these evolution trends.<br />GSM operator should consider the following documents to explicit the following evolution of IN and Network Infrastructures. <br /><br /><strong>1 New Technology Platforms </strong><br />-The views of existing IN / Internet integration taking into account the evolution of IN in terms of new technology impacts and new bearer network support.<br />-The emerging integration of IN and Internet, covering an overview of potential integration issues, such as Web-based IN Management, Web-initiated Telephony services and Web-IN, and the usage of IN principles for provision of value added services within Voice over IP networks. Furthermore this part shall provides a overview of related standards fora, such as ETSI TIPHON, IETF PINT, Parlay, JAIN, etc.<br />-Evolution of 2.5 G toward an all IP network.<br />-Applying IN technology to GPRS/UMTS for services innovation.<br />-Softswitch and other network elements for next generation network on mobile networks.<br /><br /><strong>2 New protocols</strong><br />-The impact of new object-oriented middleware technologies and new open service architectures on IN, such as Common Object Request Broker Architecture (CORBA).<br />-The impact of new service mediation technologies, such as SIP, Web services, OSA/Parlay, and JAIN.<br />-The impact of other Internet Protocol, such as H323, MGCP, H248, SIP, SIP-T. <br />-The other impacted protocols that vendors is aware. <br /> <br /><strong>3 External Rating Engine</strong><br />-GSM operator is seeking for Convergent systems that provides flexible and real-time charging and payment support to generate revenue from new 2.5G and 3G services for voice and data. The external rating engine of Convergent systems shall set up all kinds of service plans where a service charge can be calculated based on GSM Operator promotion. And it shall creates a charge and sends an already rated transaction to the GSM Operator’ s billing system where it can be included on a bill that shows all services new and old, a customer uses.<br />-The Record file that is created by external rating engine shall be processed as the event occurs. Once the event is rated and charged, the flexible output interface automatically distributes rated events to other applications, such as a legacy billing and invoicing system or even external databases or data warehouse systems. This delivers real-time feedback and immediate integration with internal and external systems.<br />-Vendors should provide the documents of external rating engine system that support the above requirement and can support the following requirements: <br />-GSM Operator can optimize the pricing for each customer and service without affecting the existing front and back office operations. <br />-The external rating engine shall enables real-time charging by capturing the real-time events from the network infrastructure across various bearers such as voice, SMS, data, content, MMS, video streaming, etc.<br />-The external rating engine shall provide the charging for prepaid, postpaid, and online payment. The rating functionality shall supports all service usage of mobile, voice, data and enables GSM Operator to support a very flexible multiservice pricing policy for both the enterprise as well as the residential market by offering tariffs based on date, time, user profile, destination or origin, all with customizable accuracy. Additionally, end users will be able at any time-before and after the transaction- to know the detailed price of a service.<br /><br /><strong>4 Service Interaction </strong><br />According to almost networks, the IN infrastructure is composed of the SSP overlay network , the Prepaid SCP, and the RBT SCP. The IN<br />system is CAMEL. And every IN call is being routed to one of the IN-SSPs and then a trigger is sent to one of the SCPs. By evolutionary extension of the service component ' s functionalities (and corresponding enhancement of the IN architecture) more advanced telecommunication services can be realized, while still supporting existing services. We have seen that the technology for service interaction with the existing IN services. For example, the new VPN service can activate with GSM Operator ‘s prepaid subscribers by trigger the existing prepaid SCPs, and it can play the RBT by trigger the existing RBT SCPs for that. The VPN service may activate different resources based on parameters such as specific service, subscriber, type of call and others. As was already described, VPN is required to interact with various existing SCPs in the network using mainly IN triggering. In order to do that, vendors should state or submit the document to GSM Operator to explicit the solutions, which provides the service interaction. And vendors should state the document that show the triggering signaling flow between the new systems and existing GSM Operator ‘s IN for this convergence service and other enhanced service. <br /><br /><strong>5 Service Provisioning and Management </strong><br />For giving these services to our subscribers, the vendors shall indicate the information of proposed system that supported the web–base serviced provisioning. The requested web-base services provisioning are detailed as follows.<br />-The systems shall provide the GUI capabilities on web base or Java base to perform fault management, configuration management, performance management, security management, file management and backup management.<br />-An services platform shall support web-base monitoring, managing, controlling, etc.<br />-The systems shall provide web-based management for remote setup, configuration file download, firmware upgrade and maintenance.<br />-The systems shall provide mechanism to ensure that only authorized GSM Operator personnel are allowed to access and manipulate the service data.<br />-The systems shall provide reports of network usage, traffic data and call blockage reports and feature activation, etc. Customer can get the reports in real time via web-base services.<br />-The systems shall support password management of various authority levels, providing detailed logs of various operations: Authority level and password protection can be set of operations such as rate management, bill management query, traffic management, service data management, user data management, access management and statistic task management query. Each operation will be documented in detailed log.<br />-The systems shall enable subscribers to activate and manage services and features either via web – base services.<br />-The systems shall provide API for connecting to GSM Operator ‘s Mediation that GSM Operator can activate and manage services and features either via mediation.<br /><br /><strong>6 Service Creation</strong><br />For giving additional services (based on convergence service systems) to our subscribers, the demand is quick customization of the ordered services, and in the requirement of rapid availability of the modified or reconfigured services. Or GSM Operator would like to have the rapid new services that developed by the third party. The vendors shall propose the solutions and submit the documents to GSM Operator for explicit solutions by the following item.<br /><br /><strong>7 Application Programming Interface (API)</strong><br />-The system shall support open APIs such as INAP, Java SIP Servlet and JAIN/Parlay to create new services. The APIs shall provide access to underlying services, switching functions and other system resources for Telco, enterprises and third-party to create their own services. <br />-The vendors shall provide details of the processing capability of the proposed network solution and shall describe the relationship between the stated capacity and calls handled.<br />-The vendors shall provide information on the behaviour of the proposed network and its call processing capability under overload conditions.<br />-The vendors shall provide the structure and details of the proposed API. As a result, GSM Operator could able to develop and implement additional services in the future ourselves.<br />-The vendors shall provide list of the partners who provide 3rd party application-servers (based on the proposed convergence service systems). The vendors shall also provide the site reference(s) that use these partners’ product. The configuration for interworking the proposed convergence service system and 3rd party application-servers shall be provided in the proposal (including the signaling protocol and call-flow).<br />-The vendors shall provide as much as possible the necessary technical information itemized in RFI regarding to GSM Operator ’s requirements.<br /><br /><strong>8 Service Creation Environment(SCE) or Software Development Kit(SDK)</strong><br />-The vendors shall provide SCE or SDK for creation, modification, customization and provision of new services on the convergence service system. SCE or SDK, which is basically based on Feature Blocks, easy to use Java-based graphical environment minimizes service-creation time and Open to third-party service developers.<br />-The vendors shall provide the details of Feature Blocks development model for fast service creation.<br />-The vendors shall provide the details of each Feature Blocks for the creation of a wide range of services.<br />-The vendors shall provide the examples of detailed services that developed by these Feature Blocks.boonsrihttp://www.blogger.com/profile/03406139995152339005noreply@blogger.comtag:blogger.com,1999:blog-7745710962466059587.post-69927775214566504912007-07-14T09:04:00.002-07:002007-07-23T07:45:18.913-07:00Call Center<strong>Call Center Feature </strong><br /><br /><strong>1. Automatic Call Attendance<br /></strong> -Any incoming calls to the main numbers will be attended by the automatic greeting with Multi Layer Voice portal of the company and shall offer the call routing function to any numbers in the company e.g. “Welcome to Service PLC – please press 1 for sales team, press 2 for accounting, press 3 for customer support, press 0 for agent or press the extension number that you know”. It shall also support concurrent attendants.<br />-Call center system shall provide difference automatic greeting and difference call routing function to any numbers in the company for each company main number.<br />-Call center system shall provide tool to each company for recorded welcome announcement or automatic greeting and managed call routing to any numbers in the company.<br /><br /><strong>2 Call Transfer<br /></strong>-Any incoming calls to the main number shall be transferred at anytime by agent and by each individual. The transferring method is open such as dialing special key or sending special commands through the installed J2ME client on the phones that are supported by most of the phones in the market (> 60%).<br />-Call center system shall generate CDR that give detailed for any incoming calls to the main number and agent or each individual doing transferred call to any numbers.<br /><br /><strong>3 Call Queuing<br /></strong>-Any incoming calls to the main number shall be queued if the agent or attendants cannot response immediately.<br />- Calls to main numbers are distributed to available attendants or agent. If no attendant (or agent) is available, calls are queued.<br />-Administrator can specify a maximum number of calls that can be queued. The number can be specified for each company separately.<br />-Administrator can specify a maximum waiting time for incoming calls. The time can be specified for each company separately.<br />-If there is no agent available to handle the incoming call, the call is queued. The caller will hear the normal alerting tones, or company specific queue messages. As soon as an agent becomes available, an attempt to connect the call that has been in queue for the longest time will be done. If no agent becomes active before the defined maximum queue time limit is exceeded, the call is handled according to the overflow plan for this scenario.<br />-When the incoming-call queue for a company has the maximum allowed number of callers, or when a call in a queue has waited longer than the specified maximum waiting time, calls are handled according to predefined directions at the following<br /> -A specified announcement is played and the incoming call is terminated.<br />-A specified announcement is played and the call is connected to another number that can be for example: A defined voicemail, another numbers.<br />-The call queue bar in Attendant Client shall show the queues by name, the number of calls in each queue, and the queuing time for the call that has been in the queue the longest.<br />-The different queues shall be assigned different priorities. There might be, for example, a high priority queue, normal priority queue and a queue for internal calls (low priority).<br /><br /><strong>4 Agent Console (or Switchboard or Attendant Client)<br /></strong>-The agent console shall support all call management through web interface/GUI or special-key phone.<br />-The attendant client shall be efficient tool for handling of incoming calls. Corporate directory and Short message handling functions are also accessed via attendant client.<br />-Attendant Clients are typically located on a customer’s premises. The Attendant Client application (HTTPs) access to Call center system via public internet.<br />-GSM Mobile and fix line telephone handset for agent voice communication is also required. Since no additional hardware is needed, the call agent can be located practically anywhere.<br />-The call agent can use the attendant client to retrieve calls from queues, park calls, transfer calls, returned calls, put callers in a queue to busy parties, initiate own calls by corporate directory or any manually entered public number, etc.<br />-Attendant Client shall provide corporate directory with include extensive search features that can be used to find relevant information. Information in the name, department, skills, title and phone number fields, among others, can be used as search criteria. Also, different ways to spell a name, or nicknames of a subscriber, can be listed in the directory and later used to find the correct information.<br />-Attendant Client shall print basic reports based on the information in the corporate directory, Call detail record of their company, Call successful and call failure of their company, etc.<br />-Attendant Client shall be able to add/modify/remove greeting menu.<br />-Any mobile of the same company can make outgoing calls via Call center system with main numbers showing.<br /><br /><strong>5 Company phone directory</strong><br />-The phone directory shall contain real-time information about the company and its employees, such as the name, department, skills, title, address, additional note and phone number. The attendants client shall have the tool to find the information or person the caller asked for.<br />-An external phone book shall be used to store contact information for external contacts with each entry including name, company, phone numbers, address, and additional note. The attendant shall have the tool to find the information or person the caller asked for.<br /><br /><strong>6 Group SMS</strong><a name="_Toc154392216"></a><a name="_Toc153972933"></a><a name="_Toc144271473"></a><br />-The user or call agent can log in to the attendance client to send SMS or group SMS. The authorized company user can use the group SMS function of the attendance client to notify the company information to the employees. The authorized company user can dial the company central number to tell the call agent to send SMS to specified numbers of groups. The user can leave a message to the Attendant, who edits it to message and sends it to the recipients. The Call center Call center system shall send the SMS to a group of users at specified time.<br /><br /><strong>7 Agent and company extension number availability status</strong><br />-The Call center system shall keep track of the states of the agents and extension number. The hunting schemes use the information when connecting calls.<br /><br /><strong>8 Hunting Schemes</strong><br />-When the incoming calls to the main number are connected to the active agents or extension number, the Call center system shall has different algorithms to select the agent or extension number for call connecting. The automatic call distribution and group hunting is based on the priorities assigned to the agents (extension number), the orderliness of list number, status of the agents (extension number) and the time agents have been idle since the last handled call.<br />-The Call center systems shall have algorithms that protect looping from automatic Call center system hunting or hunting schemes.<br /><br /><strong>9 Recording</strong><br />-Call centers in regulated industries have a legal obligation to record calls-All customer interactions must be monitored and recorded-Recorded calls are critical for resolving customer disputes-Protect confidential customer information-Recorded calls are required for training and coaching <br /><br /><strong>10 Variations on the generic call center model<br /></strong>The various components in a call center have many variations on the model developed above. A few of the variations are listed below:<br />-Remote Agents – An alternative to housing all agents in a central facility is to use remote agents. These agents work from home and use a Basic Rate ISDN access line to communicate with a central computing platform. Remote agents are more cost effective as they don't have to travel to work, however the call center must still cover the cost of the ISDN line. VOIP technology can also be used to remove the need for the ISDN, although the desktop application being used needs to be web enabled or VPN is used.<br />-Temporary Agents – Temporary agents are useful as they can be called upon if demand increases more rapidly than planned. They are offered a certain number of quarter hours a month. They are paid for the amount they actually work, and the difference between the amount offered and the amount guaranteed is also paid. Managers must use forecasting methods to determine the number of hours offered so that the difference is minimized.<br />-Virtual Call Centers – Virtual Call Centers are created using many smaller centers in different locations and connecting them to one another. The advantage of virtual call centers is that they improve service levels, provide emergency backup and enable extended operating hours over isolated call centers. There are two methods used to route traffic around call centers: pre-delivery and post-delivery. Pre-delivery involves using an external switch to route the calls to the appropriate center and post-delivery enables call centers to route a call they've received to another call center.<br />-Interaction Centers – As call centers evolve and deal with more media than telephony alone, some have taken to the term, "interaction center". Email, Web Callback, Chat and more are gradually being added to the role.boonsrihttp://www.blogger.com/profile/03406139995152339005noreply@blogger.com