VOIP Switch Monitoring and Traffic Management

VOIP Switch Monitoring and Traffic Management
Habib Madani Syed Khurram

Agenda Overview of Softswitch Voip Softswitch Call Processing Overview
Protocols Call Flows Softswitch Operations Trend analysis Summary

Overview of Softswitch
Network Overview Network Configuration Broadband Local Integrated Services Solutions

Network Overview The BTS call agent offers telephony services over IP and ATM networks. BTS provides call control intelligence for establishing, maintaining, routing, and terminating voice calls through the IP or ATM network via media gateways, while seamlessly operating with the PSTN Supports Class 5 level services, such as 911 emergency, call forwarding and caller ID Also provides support for messaging and announcements

Network Configuration
Another BTS or Call Agent MGCP SIP RUDP SS7 Links T1/Analog Lines SS7 STP BTS 10200 PSTN CO Voice Mail SERVER SS7 STP ANNOUNCE-MENT SERVER AS5300 AS5300 IP AS5300 D-Channel Backhaul ISDN PBX 3660 CAS PSAP 911 CISCO 2600 UBR

BLISS-T1 Signaling Interfaces
Network Management & Operational Support Systems SNMP, CORBA, TELNET, FTP, CLI, HTTP CA SIP-T Feature Servers FCP STP CA SS7 PRI PBX ISDN Backhaul Announcement Server Trunking Gateway Cisco SIP Voice Mail Server IMT CO IP Network

VOIP Softwitch Call Processing Overview
ISUP trunking Emergency Services Trunking gateway for operator services Voice mail services Announcement services

ISUP Trunking Gateway SS7 Links Routes offnet calls to ILEC or IXC over Inter-machine Trunks (IMTs) High port density Interconnects SS7 links b/w IP Transfer Point (ITP) and C4/5 via STPs BTS should be provisioned trunks of the same trunk group across multiple TGWs for redundancy Sigtran used to backhaul ISUP to BTS CA for call control BTS STP SCTP MGCP IMTs IAD RTP V MGX C4/C5

Trunking Gateway for 911 Services
BTS Uses Feature Group D, Operator Service (OS) signaling protocol TGW requires support of MGCP CAS “MO” package as described in “draft-foster-mgcp-cas-packages-00.txt” 2 types of 911s: enhanced 911 and basic 911 B911 has PSAP (Public Safety Answer Point) connect directly to TGW E911 has PSAP connect TGW via tandem switch 911 requires “keep the circuit up even if the caller hangs up” 911 trunks recommended connected to multiple TGWs for redundancy MGCP IAD RTP V MGX Alternate PSAP MF/CAS Trunks E911 Tandem Primary PSAP Selective Routing Database Automatic Location ID

Trunking Gateway for Operator Services
BTS Uses Feature Group D, Operator Service (OS) Signaling protocol TGW requires support of MGCP CAS “MO” package as described in “draft-foster-mgcp-cas-packages-00.txt” BTS sends Preferred Carrier Info to tandem switch to determine appropriate operation position OS does not have “keep the circuit up even if the caller hangs up” requirement as 911 OS trunks recommended connected to multiple TGWs for redundancy MGCP Tandem Switch MF/CAS Trunks RTP V IAD MGX Operator Postions (OPs)

Voice Mail Server Provides voice messaging capabilities Components
BTS IAD Provides voice messaging capabilities Components UC Applications Server VM SW resides. Terminates calls, records and replays messages, and interacts with backend servers Directory Server Stores subscriber profiles and information about which greetings are active and where greetings are located Message Server Stores and retrieves personal greetings, subscriber messages, and distribution lists2 MGCP RTP SIP V MGX UC App. Server IMAP SMTP Application Services LDAP Backend Services Directory Server Message Server Voice Mail Server

Announcement Server BTS
Instructed by BTS with MGCP to play announcement RTP to ingress MG (MTA or MGX) Audio files are stored in AS MGCP package options for AS: Script (MG requires scripting language support such as TCL) Announcement Server Deploy multiple AS for redundancy MGX MGCP V RTP IAD This # has been disconnected. Please check your # again. AS

Protocols SIP Methods MGCP Implementation MGCP Commands MGCP Modes
ISDN Channel Associated Signaling (CAS) CAS - PSAP/911

SIP Methods Consists of Requests and Responses
Requests (unless mentioned, each has a response) - REGISTER UA registers with Registration Server) - INVITE (request from a UA to initiate a call) - ACK (confirms receipt of a final response to INVITE) BYE (sent by either side to end a call) - CANCEL (sent to end a call not yet connected) - OPTIONS (sent to query capabilities) Messages contain SIP Headers and Body. Body might be SDP or an attachment or some other application **UA=User Agent (end device)**

MGCP Implementation Communication between the BTS call agent and the media gateway (MGW) is done via MGCP MGCP uses a sequence of commands and mandatory acknowledgements Commands contain a requested verb (action to be performed by endpoint) and additional parameters

MGCP Commands NotificationRequest-issued by CA instructing the MGW to watch for specific events, such as hook actions or DTMF tones on a specified endpoint (RQNT) Notify-used by the MGW to inform the CA when the requested events occur (NTFY) CreateConnection-used by the CA to create a connection that terminates at an endpoint inside the MGW (CRCX) ModifyConnection - used by the CA to change parameters associated with a previously established connection (MDCX) DeleteConnection - used by CA to delete an existing connection or by MGW when an existing connection can no longer be sustained (DLCX)

More MGCP Commands AuditEndpoint - used by CA to audit the status of the endpoint (AUEP) AuditConnection – used by CA to retrieve the parameters attached to a connection (AUCX) RestartInProgress - used by MGW to notify the CA when the gateway or a group of endpoints on the gateway are taken out of service or being placed back in service (RSIP)

ISDN Backhaul - Terminology for sending messages between CA - PBX through the AGW ISDN-Q Normal application layer messages sent to the CA over IP RUDP - Cisco proprietary protocol that makes UDP Reliable ISDN-Q.931 UDP IP Backhaul RUDP Backhaul is the process by with signaling is sent between devices through another device This implementation of ISDN or IP is native ISDN application layer (Q.931). Lower layer processing is handled by the intermediary device, in this case the AGW RUDP is the Cisco proprietary protocol that provides reliable in-order delivery for virtual connections. Is being standardized

ISDN Network Diagram IP BTS 10200
AS5300 D-Channel Backhaul MGCP ISDN PBX RUDP – Signaling and Call Setup/Teardown on the D-Channel MGCP – Voice, data, or video on the B-Channels

Channel Associated Signaling (CAS)
Inband signaling made up of tones carried on the same circuit as the call they are setting up Implemented via MGCP to support PBX connectivity Dual Tone Multi-frequency (DTMF) refers to the signaling (tones) generated when you touch a button on a push button pad MGCP supports all the DTMF/MF (signaling types: DTMF loopstart, DTMF groupstart, DTMF imstart, DTMF winkstart, MF FGD, MF imstart, MF winkstart,) CAS is required to: Support PBX interconnect and incoming CAS trunk interconnects Support Barge-In and Busy-Line Verification operator services (operator interrupt services) – Not supported release 1.0 Support PSAP/911 services

CAS - PSAP/911 911 services require support for MF signaling
PSAP operators must be able to hold the line even if caller goes on-hook

Call Flow CA to CA Call Flow – SIP RG to RG
TG - RG Using ISDN - MGCP Signaling PBX/ISDN PRI to RG Barge-In/Busy Line Verification BLV/BLI Call Flow

CA to CA Call Flow - SIP PSTN1 CA-1 CA-2 PSTN2 1. IAM 2. Invite 3. IAM
Trying 5. ACM Progress 7. ACM 8. ANM OK 10. ANM 1. Call comes into PSTN1 and the IAM is sent to Call Agent 1 (CA1) 2. CA-1 receives the IAM message, and based on its contents, and converts it into a SIP Invite with extensions message to send to CA2. The message consists of the following parts: The SIP message header The SDP message body The PSTN-MIME carrying IAM message parameters 3. CA-2 receives the Invite message and, after analyzing the contents of the message, builds an IAM and sends it to PSTN2. 4. Message R-100 is a provisional respons from CA-2 to CA-1, to notify that the call is proceeding towards the PSTN2 switch 5. CA-2 receives the ACM message from PSTN2. 6. CA-2 sends an R-183 informational message to CA-1. 7. CA-1 conveys the R-183 message as an ACM to PSTN1. 8. An ANM messages comes into CA-2 and is handled. 9. A R-200 message is generated by CA-2 upon receipt of the ANM message from PSTN2. The R-200 conveys that the INVITE action was successfully received, understood, and accepted. 10. CA1 on receipt of the R-200 generates and ANM towards PSTN1. 11.An ACK is sent from CA-1 to CA-2 confirming that CA-1 has received a final respons to an INVITE request. 12. Media flow now begins between the two Cas. 13. The release message comes into CA-1 containing the cause indicator. 14. CA-1 sends back a Release Complete. 15. CA-1 sends a BYE message to CA-2 indicating that CA-1 intends to tear down the call process. 16 & 17 CA-2 convays a release message to PSTN2 and immediately receives a RELC . 18 & 19. CA2 CA-2 sends a R-200 OK to CA-1 indicating that the call was successfully terminated and CA1 acknowledges the message. 11. ACK 12. Talking 13. REL 14. RELC 15. Bye 16. REL 17. RELC OK 19. ACK

4. RQNT (R:hu, hf, [0-9:#*T](D), S:dl)
Onnet Call – RGW to RGW Hey Call Agent, I’m going off hook BTS 10200 RG-o RG-t Hey endpoint, let me know if your subscriber hangs up 1. Off-hook 2. NTFY (O:hd) 200 OK Hey endpoint, let me know if your subscriber hangs up, hook-flash or dials digits. Oh and signal dial tone to your subscriber 3. RQNT (R:hu (N)) 200 OK 4. RQNT (R:hu, hf, [0-9:#*T](D), S:dl) Hey Call Agent, I am letting you know that I have collected digits 200 OK 5. digits Sure, but let me know if your subscriber hangs up or hook-flash 6. NTFY (O:9,1,6,3,4,2,1,2,1,2,T) 200 OK I am going to create a backwards audio path to you in case some in-band info is played by an endpoint. Endpoint ACKs with his SDP info 7. RQNT (R:hu, hf) 200 OK 8. CRCX (M:recvonly) 200 OK orig. SDP

9. CRCX (M:sendrecv orig. SDP)
Onnet Call – RGW to RGW Hey endpoint 2, you have a caller. Creating bi-directional path to you with orig. SDP info. BTS 10200 RG-o Hey end endpoint 1, I am sending you the term. endpoint SDP information RG-t 9. CRCX (M:sendrecv orig. SDP) 200 OK with term. SDP Hey, ring the phone. 10. MDCX (M:recv only, term. SDP) 200 OK Hey, play a ring back tone, the termination is ringing too. 11. RQNT (R:hd S:rg) 200 OK 12. Ringing Hey call agent, my subscriber answered the phone 13. RQNT (R:hu S:rt) 200 OK 14. Ring Back 15. Off-hook 16. NTFY (O:hd) Ok, but let me know if he hangs up 200 OK 17. RQNT (R:hu, hf) 200 OK

Bi-Directional Voice Path
Onnet Call – RGW to RGW I am making your connection bi-directional so you can talk to your buddy and hear him BTS 10200 RG-o RG-t Hey endpoint, let me know if your subscriber hangs up or hook-flashes 18. MDCX (M:sendrecv) Conversation: Aunt Pearl tells Sally about her new groovy hairstyle 200 OK Bi-Directional Voice Path 19. RQNT (R:hu, hf) Hey call agent, my subscriber went on-hook. 200 OK 20. On-hook 21. NTFY (O:hu) OK, but let me know if your subscriber goes back off-hook. Ok I am deleting the connection to you because the originating subscriber is now on-hook. 200 OK 22. RQNT (R:hd) 200 OK 23. DLCX 250 Connection Deleted

Onnet Call – RGW to RGW Deleting your connection because you went on-hook BTS 10200 RG-o Let me know if your subscriber goes on-hook. RG-t 24. DLCX 250 Connection Deleted Let me know if your subscriber goes off-hook 25. RQNT (R:hu) 200 OK 26. RQNT (R:hd) Hey my subscriber went on-hook 200 OK 27. On-hook Ok, but let me know if your subscriber goes off-hook. 28. NTFY (O:hu) 200 OK 29. RQNT (R:hd) 200 OK

TG - RG Using ISDN - MGCP Signaling
BTS 10200 ISDN Backhaul PBX MGCP/IP MGCP/IP PRI TG-2 RG-2 IP User 2 User 1

PBX/ISDN PRI to RG PBX User 1 EO/ User 2 User 3 BTS 10200 TG-1 TG-3
SETUP SETUP Backhaul CALL PROC CALL PROC CRCX (M:recvonly) ACK (SDP1) CRCX (M:sendrecv, SDP1) MDCX (M:recvonly SDP2) ACK (SDP2) ACK RQNT (R:hd, S:rg, rbk(xxx)) Alert 10. ACK Alert 12. Ringing 13. Ring back tone 14. Off-hook

PBX/ISDN PRI to RG (cont'd)
User 1 BTS 10200 EO/ User 2 User 3 TG-1 TG-3 RG-2 15. NTFY (O:hd) 16. OK 17. RQNT (R:hu) 18. ACK 19. MDCX (M:sendrecv) 20. ACK 21. CONN 22. CONN ACK 23. Bearer Connection Established

Barge-In/Busy Line Verification
Permits operators to establish a connection to a customers line to verify a busy condition Operator access is provided over dedicated facilities Facilities connect directly to a switchboard or via a switched network accessed by remote operator systems The trunks may use reverse battery loop or E&M lead supervision with multi-frequency (MF) or dial pulse (DP) signaling

RQNT (R:MS/inf,MS/rel) NTFY(O:MS/inf(digits))
BLV/BLI Call Flow BTS 10200 A-RG B -TG/RG TG Caller C Operator Conversation a Conversation seize NTFY (O:MS/sup) RQNT (R:MS/inf,MS/rel) ACK wink-start digits NTFY(O:MS/inf(digits)) CRCX (M:recvonly) ACK(SDPc) CRCX (M: inactive, SDPc) ACK(I: BLV-2, SDPa) Beaver: BLV/OI Featurette Specification EDCS a. First A party connection (BLV-1) is in sendrecv mode. b. For OI, operator plays inband alert tone and begins speaking. answer is sent as soon as possible. MDCX (M:reconly,S:MS/ans, SDPa) ACK MDCX (M: confrnce,I:BLV-1) answer ACK MDCX (M: confrnce,I:BLV-2) ACK MDCX(M:sendrec) ACK 3-way call active b Operator reports line is busy

Agenda Overview of Softswitch Voip softswitch Call Processing Overview

Softswitch Operations
Network management and Performance Counters Network Performance- Voice Quality Defining jitter, packet loss and latency VOIP Switch Performance Counters Performance Counters Flow Types of Counter on CISCO BTS Performance counter monitoring Focusing on primary Performance Monitoring.

Network management and Performance Counters
One of the key Network Management aspects is monitoring Performance counters or Performance Pegs. Performance counter collection and reporting Typically in NMS/EMS and NE the Performance data is collected as reports. Performance counters are collected in various time buckets, these buckets keep historic and pseudo-real time data. The pseudo-real time buckets can be reset for immediate trouble shooting. These reports are also periodically dumped to disk as flat files. These files are then pulled off to a data store to perform Data mining. Common Service Provider (SP) usage Preemptive trend analysis for capacity planning Service Level Agreements (SLA) Quality of service monitoring and network trouble shooting. at 5,15,30 or 60 minute intervals with switches keeping data in persistent store like a database for 24,48 hours and some times longer. Data store being a server or another storage device. Data mining is done to do trend analysis reports and capacity planning. Using for vendor SLAs .. To have 5 9s reliability. The SP achieves this by Visual monitoring of the PM counters and by reacting to Alarms/Events generated based off the set thresholds.

Network management and Performance Counters cont ..
How do we measure network performance? VOIP network Key Measurements are based of Call Success Rate , Voice Quality and Voice Mail access % Ineffective Attempts Network issues: IP backbone partially down, DNS servers partially down, voic trunk congestion, HFC/Cable plant capacity. No Channels available for Off-net PSTN calls. % Dropped Calls OR IP backbone completely down, total outage Call Processing Failure at the PSTN, signalling link is down or the bearer trunks are down. Thus all these factors are deterministic of VOIP network performance and they need to be effectively monitored. Call Success rate is defined as 1)Ineffective 2)Dropped Calls. Ip bk. Bone partially down causing congestion in call routes. VM trunks overloaded. Current capacity dedicated for voice for upstream/down bandwidth is maxing out, seen as the case in cable MSOs.

Network Performance- Voice Quality
It is dictated by Mean Opinion Score or MOS in short. How listeners perceive voice quality. Key Factors affecting Voice Quality for VOIP network: Jitter Delay in packet loss Latency We mentioned Call Success rate, Voic Access along with these factors, Voice Quality is a key VOIP Performance metric. Acceptable MOS scores for VOIP are 4 and greater. Determined by how … Equipment provided by vendors such as SAGE, can be used to determine MOS scores as well on test call basis. SPs also develop NTLPs for voice quality analysis.

Defining jitter, packet loss and latency
Delay is the time taken from point-to-point in a network. Delay can be measured in either one-way or round-trip delay. VoIP typically tolerates delays up to 150 ms before the quality of the call is unacceptable Jitter is the variation in delay over time from point-to-point. If the delay of transmissions varies too widely in a VoIP call, the call quality is greatly degraded. VOIP Network compensates for this by having jitter buffers. Packet loss is losing packets along the data path, which severely degrades the voice quality. A quick overview Jitter buffer, store packets and play them out so that the voice quality is not degraded. PSTN also introduces echo in the call path, so echo cancellation mechanisms need to be introduce on the trunking gateway vocoder cards.

Performance Counters and VOIP Switch Vendors
Current Performance Counter Availability Currently Counters are available through private Interfaces which capture the VOIP call segments. ISUP counters for PSTN signaling, SIP counters , MGCP counters for trunk gateways, general Call processing counters and QOS counters. Industry Standard for VOIP monitoring To Monitor VOIP Performance, Standard collection and polling mechanisms should available through SNMP/MIBS, CORBA/IDL, CMIP/Q3. Alerting based of the Performance Counters The VOIP switch vendors need to implement configurable thresholds mechanisms, acting as a high/low/variable water marks. These watermarks would act as triggers for alarms and events, allowing real time monitoring of the System. There is a lack of composite monitoring standard It would dictate guidelines for Performance counters, collection mechanism Alert trigger and generations. 1- But no composite open standard is defined to for these counters. 2 -VOIP Call Segments can be monitored through these counters. Some of the counters that are available .. SIP counters for SIP Trunking, Call Setup and Voic .

Performance Counters Flow
Signaling GW MG V CMS/ SoftSwitch MGC LNP STP PSTN CM NCS MTA Dqos Counters Provider Backbone HFC Plant NCS EMTA DOCSIS CMTS NCS EMTA VM CONF SRV ANN Media Servers CALEA Dqos Counters MGCP Counters SIP Counters ISUP Counters

Case Study leveraging counters available on CISCO BTS
CISCO BTS offers a wide set of performance counters through its private interfaces SNMP MIB being one of them. The following set of BTS counters capture system health across various VOIP call segments: ISDN User Part (SS7/PSTN) signaling protocol related information. MGCP signaling protocol related information. SIP Interface Adapter related information Call Processing specific information Trunk Group usage information Dynamic Quality of Service related information We present a …. There are more available.

Agenda Overview of Softswitch Voip softswitch Call Processing Overview

Case Study for Trend Analysis
Trend Analysis and Visual Monitoring of Performance Counters Architecture of DDRAW setup DDRAW Dash Board in depth Trend Analysis using Call Performance counters Trend Analysis using MGCP Performance counters Trend Analysis using ISUP Performance counters Trend Analysis using Dynamic Qos Performance counters Trend Analysis using SIP Performance counters Trend Analysis for PSTN Bearer Trunks So we are continuing with the case study using CISCO BTS for doing Trend Analysis ..

Trend Analysis and Visual Monitoring of Performance Counters
BTS performance data collection BTS performance data was collected for a USA CABLE MSO. The reports were collected at 15 min. buckets for 24 hour, and dumped to flat files (CSV). 3 months worth of these data was collected. Pull data The data was ftped over to a linux server which had Perl, Round Robin Database(RRD) and DRRAW(cgi) installed on them. RRD update Perl was used to parse the CSV files and RRD was updated with 3 months of cable MSO performance data. Multi-line Service provider

Trend Analysis and Visual Monitoring of Performance Counters (cont.)
DDRAW was customized to create a Dashboard for some of these counters. RRD and DRRAW we have a pseudo real-time display of the performance stats. This Dashboard displays past 28 hours, 1 week, month and year of data. It can be used to create a system wide view of the VOIP call flow, do capacity planning, and keep on top of SLA. Through RRD and DRRAW So you can see I keep on coming back to this visualization of the network, capacity planning and so on, .. But I have seen this to be the key for a successful operation of SP NOC.

Architecture of DDRAW setup
Client Linux Server Pull Data Pull PM Data Client View HTTP BTS 01 DRRAW CGI DRRAW CGI Apache Perl Script Parsing Parse Data Populate Data Client BTS 02 MGCP Callp RRD ISUP BTS 03

Dash Board in depth Call Stats Dash Board display, captures, the number of originating call attempts of all types, call attempts, call originating failures and call success on the reporting BTS. DQOS Dash actually looks ok, which reflects the CMTS leg, it shows that the Gate SET attempts are equal to the Gate SET Successes. MGCP Dash shows number of mgcp attempt success, fail or abandon So we have

Dash Board in depth cont ..
ISUP Dash shows SS7 signaling pattern, which includes number IAM, ANM and REL message SIP Dash shows number of SIP messages going thru the switch Trunk Dash shows utilization of trunk and overflows- Which may help in capacity planning

Dash Board

Dash Board cont.

Trend Analysis using Call Performance Counters
It can easily be seen that there is a pattern. Where the peaks represent the busy hour. Any deviations from these patterns represents anomalies that would need to be investigated by the Service Provider. Also we see a clear gap between the Success and the Call originations, this gap indicates that we are losing calls. It could be a result of hang-ups, busy dial, or network problems.

Trend Analysis using Call Stat counters – Example

Trend Analysis using Call Stat counters – Example cont.

Trend Analysis using MGCP Performance counters
It can help monitor any performance issues in the network just by reviewing ave mgcp attempts and failures. Information can be segregated by gateway which could be related to trunking gateway or announcement server.

Trend Analysis using MGCP Performance counters – Example

Trend Analysis using MGCP Performance counters – Example cont.

Trend Analysis using ISUP Performance counters
It can identify issues related to PSTN signaling, or any ss7 link problems. The Dashboard example is showing IAM,ANM,REL counters being monitored. Any anomaly related to PSTN network/usage related to incoming/outgoing PSTN calls would be visible. As an example drop in IAM would clearly indicate the call originations are having issues.

Trend Analysis using ISUP Performance counters - Example

Trend Analysis using ISUP Performance counters – Example cont.

Trend Analysis using Dynamic Qos Performance counters
COPS protocol Gate Sets attempts, Success and Commits are tracked Problem trend Network problem related to CMTS are visible through these Gate counters. A difference in Gate Set attempts and Gate Set Success would be a clear indication of CMTS resource allocation issue. A slight deviation from the norm could be an indication of a problem before SP starts experiencing it. DQOS parameters of Jitter, Latency and Packet loss are also collected by the BTS and can be monitored in this way. So we can monitor the IP or CABLE PLANT leg of the call through these counters. We can have upper/lower thresholds for these parameters, as trouble indications.

Trend Analysis using Dynamic QoS counters – Example

Trend Analysis using Dynamic QoS counters – Example cont.

Trend Analysis using SIP Performance counters
BTS uses SIP to interact with VOIC equipment and SIP trunks. Dash Board SIP Counters SIP Outgoing messages, SIP Outgoing Success,5xx Errors Problem trends An increase in retransmits or increase in 5xx errors is a visible indication that Voice mail server or SIP trunks is having issues. Voice Mail server can further be isolated by tracking the NOTIFIES.

Trend Analysis using SIP counters – Example

Trend Analysis using SIP counters – Example cont.

Trend Analysis using Trunk Usage Counters
Dash Board Trunk Counters Trunk total overflow, Incoming Trunk Busy, Outgoing trunk Busy, Total Trunk usage. A pattern is seen we can see that most of the trunk seizers are for outgoing trunks. Incoming trunk seizures are low. Overflow of trunks is very low. Problem indication Total Trunk Usage goes high, Overflow of trunks goes high are indications of capacity issue.

Trend Analysis using Trunk usage counters – Example

Trend Analysis using Trunk usage counters – Example cont.

Summary VOIP switch technology is a new field.
Performance counter aspect of network management can be a key factor in monitoring the network for issues of equipment malfunction, degradation and capacity. To provide seamless customer experience from traditional to softswitch, this will help us identifying issue proactively. A monitoring strategy is to use RRD and Drraw, for monitoring the system through pseudo real time graphs at the NOCs. Compared to the legacy Class V/IV switch technologies,

VOIP Switch Monitoring and Traffic Management

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