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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. Agenda. Overview of Softswitch Voip Softswitch Call Processing Overview
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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
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 10200 call agent offers telephony services overIP and ATM networks. • BTS 10200 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
MGCP SIP RUDP SS7 Links T1/Analog Lines Voice Mail SERVER ANNOUNCE-MENT SERVER BTS AS5300 3660 AS5300 IP AS5300 10200 CAS PSAP 911 CISCO2600 UBR ISDN PBX ISDN PBX Network Configuration Another BTS 10200 or Call Agent SS7 STP PSTN CO SS7 STP D-Channel Backhaul
BLISS-T1 Signaling Interfaces CA Network Management & Operational Support Systems SNMP, CORBA, TELNET, FTP, CLI, HTTP SIP-T Cisco FCP STP SIP Voice Mail Server PRI PBX IMT Trunking Gateway CO Announcement Server Feature Servers CA SS7 ISDN Backhaul IP Network
Agenda • Overview of Softswitch • Voip Softswitch Call Processing Overview • Protocols • Call Flows • Softswitch Operations • Trend analysis • Summary
VOIP Softwitch Call Processing Overview • ISUP trunking • Emergency Services • Trunking gateway for operator services • Voice mail services • Announcement services
STP MGCP V MGCP RTP 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 SCTP IMTs IAD MGX C4/C5
MGCP MGCP V RTP 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 IAD MGX Alternate PSAP MF/CAS Trunks E911 Tandem Primary PSAP Selective Routing Database Automatic Location ID
MGCP MGCP V RTP 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 Tandem Switch MF/CAS Trunks IAD MGX Operator Postions (OPs)
MGCP MGCP SIP RTP V SMTP LDAP IMAP Voice Mail Server BTS • 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 IAD MGX UC App. Server Application Services Backend Services Directory Server Message Server Voice Mail Server
MGCP MGCP V RTP This # has been disconnected. Please check your # again. 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 IAD AS
Agenda • Overview of Softswitch • Voip Softswitch Call Processing Overview • Protocols • Call Flows • Softswitch Operations • Trend analysis • Summary
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 10200 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 toinform 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 toretrieve 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)
Backhaul ISDN-Q.931 RUDP UDP IP ISDN • Backhaul- Terminology for sending messages between CA - PBX through the AGW • ISDN-Q.931- Normal application layer messages sent to the CA over IP • RUDP- Cisco proprietary protocol that makes UDP Reliable
BTS AS5300 IP 10200 ISDN PBX ISDN PBX ISDN Network Diagram D-Channel Backhaul MGCP 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 samecircuit 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
Agenda • Overview of Softswitch • Voip Softswitch Call Processing Overview • Protocols • Call Flows • Softswitch Operations • Trend analysis • Summary
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
1. IAM 3. IAM 2. Invite 4. 100 Trying 6. 183 Progress 5. ACM 7. ACM 8. ANM 9. 200 OK 10. ANM 12. Talking 13. REL 14. RELC 15. Bye 18. 200 OK 16. REL 17. RELC 19. ACK CA to CA Call Flow - SIP CA-1 CA-2 PSTN1 PSTN2 11. ACK
RG-o RG-t 8. CRCX (M:recvonly) 200 OK orig. SDP 200 OK 200 OK 200 OK 200 OK 200 OK BTS 3. RQNT (R:hu (N)) 7. RQNT (R:hu, hf) 4. RQNT (R:hu, hf, [0-9:#*T](D), S:dl) 10200 Onnet Call – RGW to RGW Hey Call Agent, I’m going off hook Hey endpoint, let me know if your subscriber hangs up 916-342-1206 916-342-1212 1. Off-hook 2. NTFY (O:hd) Hey endpoint, let me know if your subscriber hangs up, hook-flash or dials digits. Oh and signal dial tone to your subscriber Hey Call Agent, I am letting you know that I have collected digits 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) 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
RG-o RG-t 12. Ringing 14. Ring Back 200 OK 200 OK 200 OK with term. SDP 200 OK 200 OK 200 OK BTS 10. MDCX (M:recv only, term. SDP) 10200 16. NTFY (O:hd) 15. Off-hook Onnet Call – RGW to RGW Hey endpoint 2, you have a caller. Creating bi-directional path to you with orig. SDP info. Hey end endpoint 1, I am sending you the term. endpoint SDP information 916-342-1212 9. CRCX (M:sendrecv orig. SDP) Hey, ring the phone. 11. RQNT (R:hd S:rg) Hey, play a ring back tone, the termination is ringing too. Hey call agent, my subscriber answered the phone 13. RQNT (R:hu S:rt) Ok, but let me know if he hangs up 17. RQNT (R:hu, hf)
RG-o RG-t 200 OK 200 OK 200 OK 200 OK BTS 18. MDCX (M:sendrecv) 10200 21. NTFY (O:hu) 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 Hey endpoint, let me know if your subscriber hangs up or hook-flashes 916-342-1212 Conversation: Aunt Pearl tells Sally about her new groovy hairstyle 19. RQNT (R:hu, hf) Hey call agent, my subscriber went on-hook. 20. On-hook 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. 22. RQNT (R:hd) 23. DLCX 250 Connection Deleted
RG-o RG-t 250 Connection Deleted 200 OK 200 OK BTS 10200 28. NTFY (O:hu) Onnet Call – RGW to RGW Deleting your connection because you went on-hook Let me know if your subscriber goes on-hook. 916-342-1212 24. DLCX 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 27. On-hook Ok, but let me know if your subscriber goes off-hook. 29. RQNT (R:hd) 200 OK
RG-2 TG-2 PBX BTS 10200 User 2 User 1 TG - RG Using ISDN - MGCP Signaling ISDN Backhaul MGCP/IP MGCP/IP PRI IP endpoint/1@tg1.cisco.com endpoint/1@rg2.cisco.com
EO/ User 2 PBX User 1 TG-1 RG-2 User 3 TG-3 SETUP SETUP Backhaul CALL PROC CALL PROC CRCX (M:recvonly) CRCX (M:sendrecv, SDP1) RQNT (R:hd, S:rg, rbk(xxx)) ACK (SDP1) BTS 10. ACK ACK (SDP2) 10200 MDCX (M:recvonly SDP2) ACK Alert Alert 12. Ringing 13. Ring back tone 14. Off-hook PBX/ISDN PRI to RG
EO/ User 2 PBX User 1 TG-1 RG-2 User 3 TG-3 16. OK 17. RQNT (R:hu) 19. MDCX (M:sendrecv) 23. Bearer Connection Established BTS 20. ACK 15. NTFY (O:hd) 18. ACK 21. CONN 10200 22. CONN ACK PBX/ISDN PRI to RG (cont'd)
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
A-RG TG Operator B -TG/RG Caller C a Conversation Conversation seize wink-start RQNT (R:MS/inf,MS/rel) NTFY (O:MS/sup) digits ACK BTS NTFY(O:MS/inf(digits)) 10200 CRCX (M:recvonly) ACK(SDPc) CRCX (M: inactive, SDPc) ACK MDCX (M: confrnce,I:BLV-1) ACK(I: BLV-2, SDPa) answer ACK ACK MDCX(M:sendrec) b 3-way call active BLV/BLI Call Flow MDCX (M:reconly,S:MS/ans, SDPa) MDCX (M: confrnce,I:BLV-2) ACK Operator reportsline is busy
Agenda • Overview of Softswitch • Voip softswitch Call Processing Overview • Protocols • Call Flows • Softswitch Operations • Trend analysis • Summary
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
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.
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, voicemail 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.
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
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.
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.
Signaling GW MGC STP V LNP CMS/ SoftSwitch CM NCS MTA Dqos Counters NCS EMTA DOCSIS MG CMTS NCS EMTA PSTN HFC Plant Provider Backbone ISUP Counters SIP Counters MGCP Counters Dqos Counters ANN SRV CONF SRV VM CALEA Media Servers Performance Counters Flow
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
Agenda • Overview of Softswitch • Voip softswitch Call Processing Overview • Protocols • Call Flows • Softswitch Operations • Trend analysis • Summary
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
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
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.