The Sip School and Cisco Web in Ar

2011 Cisco and/or its affiliates. All rights reserved. 1

Design & Implementation of SIP Trunking using Ciscos Session Border Controllers

Graham Francis CEO, The SIP School

Darryl Sladden Technical Marketing Manager, Cisco

Pashmeen Mistry Technical Marketing Engineer, Cisco

October 27th 2011


Founded in April 2000

5300+ Students

Provide the Industry recognised SSCA SIP Certification program, endorsed by the TIA + more.

eLearning in modular format

Unique as content evolves as SIP evolves

Connected with Cisco to provide SIP foundation training

http://cisco.thesipschool.com / Discount codes later.

Now, lets talk about why were all here today and well start with SIP


6644 55

9977 88

##** 00

VideoVideo

3311 22

HoldHold

6644 55

9977 88

##** 00

VideoVideo

3311 22

HoldHold

SIPSupport Video?

SIP OK

Call 1003

VOICE MEDIA

1003 OK 1002 OKSIPWant to talk?

SIP OK

VIDEO MEDIA


Data from

Frost & Sullivan

Now $7.44 billion by 2017


Voice Communications

Less Money

Equal / Better quality

Greater functionality


Worldwide Phenomenon

Will happen

One day, no PSTN

It is Easy to implement


Unified Clients

Unified Server

Inc.Registrar and

Location services

SIP IP Phones

DirectoryDNS

Messaging Server

GatewayPBX

Firewall / NAT

ITSP

ldap

naptr

sip

sip

sip sip sip

sip

sipSip trunk


Internet ISP

Data

Asymmetric DSL

TDM / PBX

ITSP

TDM to SIP/RTP

Gateway

SIP Trunks


Data

TDM / PBX

TDM to SIP/RTPGateway

Voice

Internet ISP

ITSP

Switch

SIP / PBX

IP Network


The road to compatibility


ITSP

Network

Your PBX

SIP Registrar

G.711 G.729G.711 to G.729

Media

SIP Signaling

REGISTER

Secured

SBC

B2BUA

SBC

B2BUA


[email protected]


Changing Landscapes VoIP Islands to VoIP Interconnects

Unified communications SIP Trunks to destinations beyond the Enterprise

IPA

IPA

Enterprise Domain 1 Enterprise Domain 2

Narrowband voice to Rich-media Interconnect

A A

Enterprise Domain 1 Enterprise

Domain 2SP VoIPSBC SBCCUBE CUBE

Extend rich-media collaboration to vendors, partners and customers

A Cisco Unified Border Element (CUBE) provides b2b interconnectivity for secure rich-media services

IP IP

Enabling Business-to-Business Collaboration


Capture a 53% cost savings opportunity

Avg.

-40%


A

ACVP

Branch Offices

Campus Contact Center

A

ACVP

SP SIP

A

ACVP

SP SIP

1. TDM Trunking Yesterday

2. TDM and IP Trunking Today

3. IP Trunking TomorrowCampus Contact Center

Campus Contact Center

Branch Offices

Branch Offices


I have multiple PBXs that all need to have SIP Trunking enabled in order to get the best Return on Investment (ROI).

I would like to centralize all of my SIP Trunking in a single location.

SIP Trunking is complex new technology, how do I make Trouble shooting easier.

How can I ensure that I am compliant with my companys security policies when I implement SIP Trunking ?

Challenge Impact of an SBC

Allows you to have a single interconnect point to your Service Provider across multiple disparate systems.

Allows you to scale your SIP Trunk solution while only connecting to one device.

Allows a single point of troubleshooting for your SIP Trunks. A device that is supported by Cisco allows you to have one vendor support your entire solution.

SBCs ensures security on SIP Trunks. An SBC from a trusted vendor such as Cisco incorporates security in all aspects from an embedded firewall to administrative control on changes.

Features of a Cisco SBC

mrf4 New oneMike Fratesi, 14/05/2008


Overview


An Integrated Network Infrastructure Service

VXML

SRSTRSVP Agent

Cisco Unified Border Element

Address Hiding

H.323 and SIP interworking

DTMF interworking

SIP security

Transcoding

Unified CM Conferencing and

Transcoding

GK

TDM Gateway

Voice and Video TDM Interconnect

PSTN Backup

Routing, FW, IPS, QoS

WAN Interfaces

Note: An SBC appliance wouldhave only these features

CUBE

Note: Some features/components may require additional licensing


2800 ISR

3800 ISR

2900 ISR G2

AS5000XM

ASR 1004/6 RP2

Active Voice Call (Session) Capacity

C

P

S


CUBE Session Capacity Summary

Platform CUBE SessionsC880/C890 SKUs 5-25

1861 5-15

2801 55

2811 110

2821 200

2851 225

3825 400

3845 500

AS5000XM 600

2901 100

2911 200

2921 400

2951 600

3925 800

3945 950

3925E 2100

3945E 2500

ASR1002/1004/1006 RP1 1750

ASR1001 10000

ASR1004/1006 RP2 16000

ASR1001 introduced in RLS 3.2 in Nov 2010

Introduced in March 2011

Reference

End of Life PlatformsLast IOS Release:

15.1.4M


Platform Single-Use LicensesActive-Standby B2B

Redundancy Licenses

Cisco 2901, 2911, 2921 ISR G2

FL-CUBEE-5

FL-CUBEE-25

FL-CUBEE-100

FL-CUBEE-5-RED

FL-CUBEE-25-RED

FL-CUBEE-100-RED

Cisco 2951, 3925 ISR G2

FL-CUBEE-5

FL-CUBEE-25

FL-CUBEE-100

FL-CUBEE-500

FL-CUBEE-5-RED

FL-CUBEE-25-RED

FL-CUBEE-100-RED

FL-CUBEE-500-RED

Cisco 3945, 3925E, 3945E ISR G2

FL-CUBEE-5

FL-CUBEE-25

FL-CUBEE-100

FL-CUBEE-500

FL-CUBEE-1000

FL-CUBEE-5-RED

FL-CUBEE-25-RED

FL-CUBEE-100-RED

FL-CUBEE-500-RED

FL-CUBEE-1000-RED

Cisco ASR1000

FLASR1-CUBEE-100P

FLASR1-CUBEE-500P

FLASR1-CUBEE-1KP

FLASR1-CUBEE-4KP

FLASR1-CUBEE-16KP

FLASR1-CUBEE-100R

FLASR1-CUBEE-500R

FLASR1-CUBEE-1K-R

FLASR1-CUBEE-4K-R

FLASR1-CUBEE-16KR

More info in the CUBE Ordering Guide: http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/order_guide_c07_462222.html

Reduced Pricing for redundancy


Advanced Features


Network-based Media

Recording Solution

Business to Business

Telepresence

SIP Trunks for PSTN Access

IVR Integration for Contact

Centers

SIP

H.323

SP VOIPServivces

SIP Trunk

SBC

TDM

CUBE

SP IP Network

SIP

SBCCUBE

SIP

MediaSense

RTP

RTP

Partner API

CUBE

SIPSP IP

NetworkSBC

CVPvXML Server

Media Server

A SP IP Network

SIP SIP ASBC

CUBECUBE


Telecommuter/SOHO

V V V V

SP IP Network

TDM-based PSTN

Class 4/5 Switch

MPLS

Campus HeadquartersBranch Office

TDM Trunk Call Path

Voice

Voice

Voice

IP Trunk Call Path

Voice

Voice

CUBE

MPLS

Data Data

Data

VPNVPN


Characteristics of Centralized Operational Benefits Challenges

Central Site is the only location

with SIP session connectivity to

IP PSTN

Voice services delivered to

Branch Offices over the

Enterprise IP WAN (usually

MPLS)

Media traffic hairpins through

central site between SP and

branches

Centralizes Physical

Operations

Centralizes Dial-Peer

Management

Centralizes SIP Trunk

Capacity

Increased campus and branch bandwidth, CAC, latency; media optimization

HA in campus (single point of failure)

Survivability (backup branch call processing)

Emergency services

Legal/Regulatory, Geographical

Site-SP Media

Centralized

A

IP PSTN

Enterprise

IP WAN

CUBE


Telecommuter/SOHO

V V V V

SP IP Network

TDM-based PSTN

Class 4/5 Switch

MPLS

Campus HeadquartersBranch Office

MPLS

DataData

Data

VPNVoice

Voice

IP Trunk Call Path

Voice

CUBECUBE

VPN


Characteristics of

Distributed

Operational Benefits Challenges

Each site has direct connection

for SIP sessions to SP

Takes advantage of SP session

pooling, if offered by SP

Media traffic goes direct from

each branch site to the SP

Leverages existing branch

routers

No media hair-pinning thru any

site.

Lower latency on voice or video

Built-in Redundancy strategy

Quickest transition from existing

TDM

Distributed dial-peer management

Distributed operational overhead

Site-SP Media

A

Distributed

Enterprise

IP WANCUBE

CUBECUBECUBECUBE

IP PSTN


Characteristics of Hybrid Benefits

Connection to SP SIP service is determined on a site

by site basis to be either direct or routed through a

regional site.

Decision to route call direct or indirect based on

various criteria

Media traffic goes direct from site to SP or hairpins

through another site, depending on branch

configuration.

Adaptable to site specific requirements

Optimizes BW use on Enterprise WAN

Adaptable to regional SP issues

Built-in redundancy strategy

CUBE

Hybrid

A

CUBE

A

CUBE

Enterprise

IP WANCUBE

IP PSTN

Site-SP Media


Cisco Interoperability Portal:www.cisco.com/go/interoperability

Validated with service providers world-wide

Tested with 3rd party PBXs

Standards based


[email protected]


Digital/Analog Trunks

SIP/H.323 Trunks

dial-peer voice 2 voipdestination-pattern 9Tsession protocol sipv2session target ipv4:codec g711ulaw

dial-peer voice 2 potsdestination-pattern 9Tport 0/0/0:23

dial-peer voice 1 voipdestination-pattern 9Tsession protocol sipv2session target ipv4:codec g711ulaw

dial-peer voice 1 voipdestination-pattern 1...session protocol sipv2session target ipv4:codec g711ulaw

x 1001

CUBE

SBC

SIP TrunksSIP/H.323 Trunks

x 1001

SIP SP

Change POTS

Call Leg to

VoIP Call Leg

Re-purpose your existing Cisco Voice Gateways as Ciscos Session Border Controller Cisco Unified Border Element (CUBE)

Buy CUBE License

Only


IP

CUBE

CUBE

Actively involved in the call treatment, signaling and media streams

SIP B2B User Agent

Signaling is terminated, interpreted and re-originated

Provides full inspection of signaling, and protection against malformed and malicious packets

Media is handled in two different modes:

Media Flow-Through

Media Flow-Around

Digital Signal Processors (DSPs) are required for transcoding (calls with dissimilar codecs)

IP

Media Flow-Around

Signaling and media terminated by the Cisco Unified Border Element

Media bypasses the Cisco Unified Border Element

Media Flow-Through

Signaling and media terminated by the Cisco Unified Border Element

Transcoding and complete IP address hiding require this model


CUBE

x1001+1 408-526-6855

INVITE /w SDPINVITE /w SDP

100 TRYING

100 TRYING

10.1.1.1 20.1.1.1

c= 192.168.1.50m=audio abc RTP/AVP 0

c= 20.1.1.1m=audio xxx RTP/AVP 0

180 RINGING

180 RINGING

200 OK200 OK

c= 20.1.1.2m=audio uvw RTP/AVP 0c= 10.1.1.1

m=audio xyz RTP/AVP 0

RTP (Audio)

ACK ACK

192.168.1.50 10.1.1.1 20.1.1.1 20.1.1.2

192.168.1.1

192.168.1.50

SIP SP

SBC

20.1.1.2

Internal Network

ExternalNetwork

B2B User Agent


SIP/H.323Protocol Stack

SIP/H.323 Protocol Stack

Ingress I/F Egress I/FHW LAN/WAN Interfaces

IOS Infrastructure (ACLs, FW, IPS, VPN)

TCP UDP TLS TCP UDP TLSDSP Hardware

DSP API

DTMF xlationCodec FilteringXcoding Control

Dial-peer Dial-peer

Voice Application CodeL7 Protocol-independent memory structures holding call

state and attributes (CLID, Called #, CodecM)

Signaling

RTP LibraryRTP Library

Media

Signaling

Packets

Physical Interfaces

IOS Infrastructure

TCP/UDP/TLS Voice

stack

SIP/H323 Protocol Stack

Dial-Peer

Voice Application Code

Media

Packets

Physical Interfaces

IOS Infrastructure

RTP Library

DSP (If invoked)

2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 38

Step 0 Configure IP PBX to route calls to the edge SBC

Step 1 Get SIP Trunk details from the Provider

Step 2 Turn CUBE Application ON on Cisco routers

Step 3 Configure Call routing on CUBE (Incoming & Outgoing Dial-Peers)

Step 4 Normalize SIP messages to meet SIP Trunk Providers requirements

Step 5 Execute the Test Plan


Configure CUCM to route calls to CUBE via a SIP/H323 Trunk

Make sure all different patterns of calls local, long distance, international, emergency, informational etc.. are pointing to CUBE

SP IP Network

SIP

SBCCUBE

SIP Trunk pointing to CUBE

SIP


Item SIP Trunk service provider requirement Sample

Response

1 SIP Trunk IP Address (Destination IP Address for INVITES) 20.1.1.2

2 SIP Trunk Port number (Destination port number for INVITES)

5060

3 SIP Trunk Transport Layer (UDP or TCP) UDP

4 Codecs supported G711, G729

5 Fax protocol support T.38

6 DTMF signaling mechanism RFC2833

7 Does the provider require SDP information in initial INVITE (Early offer required)

Yes

8 SBCs external IP address that is required for the SP to accept/authenticate calls (Source IP Address for INVITES)

20.1.1.1

9 Does SP require SIP Trunk registration for each DID? If yes, what is the username & password

No

10 Does SP require Digest Authentication? If yes, what is the username & password

No


1. Turn CUBE Application ON1. Turn CUBE Application ON

2. Global settings to meet SPs requirements and SIP Trunk towards SP if needed2. Global settings to meet SPs requirements and SIP Trunk towards SP if needed

voice service voip

mode border-element license capacity 200

allow-connections sip to sip

voice service voip

sip

early-offer forced

header-passing error-passthru

midcall-signaling passthru

voice service voipip address trusted list

ipv4 10.1.1.50ipv4 20.20.20.20

3. Create a trusted list of IP addresses to prevent toll-fraud3. Create a trusted list of IP addresses to prevent toll-fraud


Dial-peer static routing table mapping phone numbers to interfaces or IP addresses

SP IP Network

SIPH.323 or SIP

SBCCUBE

INBOUND & OUTBOUND CALLS

LAN Dial-Peers

WAN Dial-Peers

LAN Dial-Peers Dial-Peers that are facing towards the IP PBX for sending & receiving calls to & from the PBX

WAN Dial-Peers Dial-Peers that are facing towards the SIP Trunk Provider for sending & receiving calls to & from the provider


dial-peer voice 100 voipdescription *** LAN side dial-peer ***incoming called-number 9T session protocol sipv2destination-pattern [2-9].........voice-class sip bind control source gig0/0voice-class sip bind media source gig0/0session target ipv4:codec g711ulawdtmf-relay rtp-nte

CUCM sending 9 + All digits dialed

SP will be sending 10 digits inbound

INBOUND DP FOR CALL FROM CUCM TO CUBEOUTBOUND DP FOR CALLS FROM CUBE TO CUCM

Note: If more than 1 CUCM exists, you will have to create multiple such LAN dial-peers with preference CLI for CUCM redundancy


Catch-all for all SP inbound calls

dial-peer voice 201 voipdescription *** WAN side dial-peer_Long distance***translation-profile outgoing Digitstrip_9destination-pattern 91[2-9].........session protocol sipv2voice-class sip bind control source gig0/1voice-class sip bind media source gig0/1session target ipv4:dtmf-relay rtp-ntecodec g729r8

dial-peer voice 200 voipdescription *** WAN side Incoming DP ***incoming called-number [2-9].........session protocol sipv2dtmf-relay rtp-nte

INCOMING WAN DIAL-PEER FOR CALLS FROM SP TO CUBE

OUTGOING WAN DIAL-PEER FOR CALLS TO SP FROM CUBE

DP for sending long distance calls to SP

Note: Separate outgoing DP to be created for Local, International, Emergency, Informational calls etc. Thus, for WAN Inbound & Outbound DP are separate


voice class sip-profiles 400

request INVITE sip-header Diversion modify sip:(.*>) sip:[email protected]>

request REINVITE sip-header Diversion modify sip:(.*>) sip:[email protected]>

request ANY sip-header User-Agent modify User-Agent:(.*) User-Agent: Cisco CUCM8.5/IOS-15.1-3

response ANY sip-header Server modify Server:(.*) Server: Cisco CUCM8.5/IOS-15.1-3

dial-peer voice 4000 voip

description Incoming/outgoing SP

voice-class sip profiles 400

Received: INVITE sip:[email protected]:5060 SIP/2.0SSSUser-Agent: Cisco-CUCM8.5

SSSDiversion: ;privacy=off;

reason=unconditional;screen=yesSS...m=audio 6001 RTP/AVP 0 8 18 101a=rtpmap:0 PCMU/8000SS...

Configure

SIP Profiles

Apply to

Dial-peer or

Globally

See the

difference

Sent: INVITE sip:[email protected]:5060 SIP/2.0SSS.User-Agent: Cisco CUCM8.5/IOS-15.1-3

SSS.Diversion: ;

privacy=off;reason=unconditional;screen=yesSSS.m=audio 32278 RTP/AVP 18 8 101a=rtpmap:0 PCMU/8000SSS..

voice service voip

sip

sip profiles 1000

SIP Provider

Requirement

1. For Call Forward & Transfer scenarios back to PSTN, the Diversion header should match the

registered DID of your network

2. The User-Agent field in all SIP messages should state the version of PBX and of SBC that is being

used


Inbound and outbound Local, Long distance, International calls for G711 & G729 codecs

Outbound calls to information and emergency services

Caller ID and Calling Name Presentation

Supplementary services like Call Hold, Resume, Call Forward & Transfer

DTMF Tests

Fax calls T.38 and fallback to pass-through (if option available)


CUBE# show call active voice brief

121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:2 Answer 2000 activedur 00:00:14 tx:0/0 rx:0/0IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a

121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:1 Originate 1000 activedur 00:00:14 tx:0/0 rx:0/0IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a

Telephony call-legs: 0SIP call-legs: 2H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 2

CUBE# show voip rtp connections

VoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP1 17 18 17474 6000 10.10.10.10 1.1.1.1 2 18 17 17476 6001 20.20.20.20 2.2.2.2

Found 2 active RTP connections


Is CUBE Active ? show cube status

Is the call matching

right Dial-peers ?

Are we sending the

right SIP call to SP based

on their requirements ?

debug voip ccapi inout

debug ccsip messages

CUBE-Version : 9.0SW-Version : 15.2.1T, Platform 2911HA-Type : noneLicensed-Capacity : 200

Oct 26 18:59:01.146: //-1/66A6B1BF8013/CCAPIcc_api_call_setup_ind_common:.................

Incoming Dial-peer=1, Progress Indication=NULL(0), Calling IE Present=TRUE,.................

Outgoing Dial-peer=100, Params=0x26E8574, Progress Indication=NULL(0)

Received:INVITE sip:[email protected]:5060 SIP/2.0Date: Wed, 26 Oct 2011 18:59:01 GMTAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYFrom: "Paul Hewson" ;tag=90d94d92-6ee4-45aa-9f18-2d09025c1ee4-27352390................


SP IP Network

SIPH.323 or SIP

SBCCUBE

SNMP Response

SNMP Query

Network Management Tools can be used to monitor key CUBE statistics like SIP Trunk status, Trunk utilization, Call Arrival Rate, Call Success/Failure count, voice quality metrics etc..

Network Management Tools can send SNMP Queries to CUBE

CUBE responds to the SNMP queries with real time values of the monitored objects

CUBE can also send SNMP Traps to alert the network management tool of certain events like SIP Trunk failure, link down, high CPU etc.. Network

Management Tool

Some Network Management Tools:- Cisco Unified

Operations Manager

- Arcana Networks- Solarwinds

Some Network Management Tools:- Cisco Unified

Operations Manager

- Arcana Networks- Solarwinds


Area Information Method

Router Health CPU, Memory, I/f CISCO-PROCESS-MIB, cpmCPUTotal5minRev

CISCO-MEMORY-POOL-MIB, ciscoMemoryPoolTable

IF-MIB, IfEntry

SIP Trunk Status SIP Trunk Status SIP OOD Options Ping, CLI dial-peer status

Traffic Reports (Calls, Sessions, Capacity Planning, Errors)

Trunk Utilization

CUBE 1.4: CISCO-VOICE-DIAL-CONTROL-MIB, cvCallVolume

Older CUBE: DIAL-CONTROL-MIB, callActive

CISCO-DIAL-CONTROL-MIB, cCallHistoryTable

CUBE 8.5: SIP RAI Trunk Utilization

Call Arrival Rate CUBE 1.4: CISCO-VOICE-DIAL-CONTROL-MIB, cvCallRateMonitor

Call Success/Failure

DIAL-CONTROL-MIB, dialCtlPeerStatsSuccessCalls, dialCtlPeerStatsAcceptCalls, dialCtlPeerStatsFailCalls, dialCtlPeerStatsRefuseCalls

CISCO-SIP-UA-MIB, cSipStatsErrClient, cSipStatsErrServer, cSipStatsGlobalFail

SIP retries CISCO-SIP-UA-MIB, cSipStatsRetry

Media Resources (DSPs)

DSP Availability CISCO-DSP-MGMT-MIB, cdspCardResourceUtilization, cdspDspfarmUtilObjects

Transcoding util. CUBE 1.4: CISCO-DSP-MGMT-MIB, cdspTotAvailTranscodeSess,

cdspTotUnusedTranscodeSess

MTP utilization CUBE 1.4: CISCO-DSP-MGMT-MIB, cdspTotAvailMtpSess,

cdspTotUnusedMtpSess

Voice QualityLoss, delay, jitter CISCO-VOICE-DIAL-CONTROL-MIB, cvVoIPCallActiveTable

IP SLA CISCO-RTTMON-RTP-MIB, rttMonJitterStatsTable , rttMonLatestJitterOperTable

More info in CUBE Management and Manageability Specification at:http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/white_paper_c11-613550.html

Reference


! create profileip traffic-export profile TAC mode capture

bidirectionalincoming access-list 123outgoing access-list 123

!! access-list to filter only SIP messages (port 5060) access-list 123 permit udp any any eq 5060access-list 123 permit tcp any any eq 5060!! apply to an interface, default memory is 5Minterface fa0/0

ip traffic-export apply TAC [size ]

router#traffic-export interface fa0/0 clearrouter#traffic-export interface fa0/0 start

router#traffic-export interface fa0/0 stop

2. Capture traffic with these exec (enable) level commands

1. Configure capture profile

IP Traffic Capture: http://www.cisco.com/en/US/docs/ios/12_4t/12_4t11/ht_rawip.html

3. Export the pcap file to a server

router#traffic-export interface fa0/0 copy ftp://x.x.x.x/capture.pcap

4. Display ladder diagram (with Wireshark)

Note: The exec cmds dont appear until a profile has been configured

Note: Allows filtering of calling/called numbers when creating the flow graph


High Availability with Inbox & Box to Box Redundancy Resiliency by alternative routing of INVITEs Media Forking for recording of calls Media Enhancements through DSPs such as Noise

Reduction and Acoustic Shock Prevention Video Call Handling via CUBE Additional Audio Codecs such as G722 and wide-band

codecs Additional SIP messages handled via Trunking specifically

Presence Indicators SIP Trunks to Webex for Cloud Connected Audio

ShippingShipping

PlannedPlanned


Visit http://www.cisco.com/go/cube for more information on CUBE

Email [email protected] for any CUBE related questions

Visit http://cisco.thesipschool.com/webinar

To get access to the Webinar

To send any comments and feedback

For discounted SSCA SIP training

To get a copy of a working config

Email [email protected] for your discount voucher.

This offer is open to webinar attendees for a limited time only.

Documents

The Sip School and Cisco Web in Ar