SIP Trunks for
PSTN Access
BRKUCT-2001
Agenda
Technology Overview
Deployment Scenarios
Issues with SIP Trunks for PSTN Access
What's in a SIP Trunk?
What services should customers look for in SIP Trunks?
What services should Service Providers offer with SIP Trunks?
Using the Cisco Unified Border Element (formerly Cisco Multiservice IP-to-IP Gateway) as a Border Element between Service Provider and Customer
Issues and configuration options for Cisco Unified Communications Manager Express (CUCME) with SIP Trunks for PSTN Access
Issues and configuration options for Cisco Unified Communications Manager (CUCM) with SIP Trunks for PSTN Access
General Troubleshooting Issues with SIP Trunks
SIP Trunks for PSTN Access
SIP Trunks can be defined as a method to interconnect different
SIP-based networks. SIP Trunks can be used by a wide range of organizations as a method of interconnections. One large application for SIP Trunks is for service providers to offer PSTN interconnects for voice-enabled SIP applications over an IP connection. This gives service providers the ability to offer PSTN services over a combined IP infrastructure, reducing the cost and complexity of the network and providing a single point of
interconnect to their users. It also allows them to offer services outside of their geographic regions that have a PSTN footprint and consolidate the interconnect to the PSTN across multiple customers. The benefit for enterprises and smaller businesses is that they can get PSTN
interconnect services without needing a PSTN facility at their location. This allows the removal of the PSTN hardware from their equipment and allows connections to the PSTN over a wide variety of mediums supporting IP such as wireless and cable. This session discusses implementation issues and requirements when using SIP Trunks for PSTN access.
H.323/SIP Trunk
Objectives and Scope of This Seminar
Discuss issues to consider when adding SIP Trunks to solutions Focus will be on deployments with Cisco Unified Communications
Manager and Cisco Unified Communications Manager Express Specific Service Provider offering will NOT be evaluated
Solution will focus on recommended design solution of using the Cisco Unified Border Element (formerly Cisco Multiservice IP-to-IP Gateway) as a Border Element between SP offering and customer. A Cisco UBE could be provided by the Service Provider and one should be owned by the customer
Cisco Unified Border Element
SIP Trunk Cisco Unified Communications
Manager or Cisco Unified Communications Manager Express
Service Provider
Service Provider Owned Enterprise Owned
Cisco Unified Border Element
Enterprise Located Equipment
H.323/SIP Trunk
Technology
Overview
“SIP Trunks”—Why ?
SIP Trunks can be
cheaper
(sometimes)
SIP Trunks can be more versatile (i.e. deployed over
different physical layer links)
SIP Trunks can offer equipment consolidation
SIP Trunks can be used for many different purposes
Between applications (i.e. Conference bridges to IP PBX, such as MeetingPlace® to Cisco Unified Communications Manager)
For PSTN Access (Centralized or Distributed)
Between different IP Communication Zones within a
company (i.e. Europe and USA) or between Companies (i.e. Disney and Apple)
SIP Trunks—Myth vs. Reality
Myth Reality
SIP Trunks can be deployed over any media SIP Trunks should only be deployed over media that can provided a guarantee QoS that is
acceptable (i.e. it would not be recommended to deploy them across Satellite links if Voice quality is important)
SIP Trunks are always cheaper then PSTN trunks for PSTN Access
Large Enterprise have such low rates for
traditional TDM based telephony, rates over SIP Trunks may not save that much in per minute charges for Local or Long Distance voice calls SIP Trunks provide the exact same experience
for the end users
SIP Trunks can provide the same experience In many cases, but some cases (i.e. Baudot
connections for Deaf users or V.92 speed modem connections) experience is different SIP Trunks are easy to deploy and just work SIP is easy to deploy, but interconnection
between different vendors implementations of SIP and different Service Providers offering is not yet ironed out
SIP Trunks should always be used In some cases H323 trunks make a better choice and in some cases TDM trunks make a better choice
SIP Trunks—Good Reasons for
Implementing Them
SIP Trunks offer a roadmap to
Enhanced Services
WideBand Codecs
Calls with SUBJECT lines
Exchange of Calendaring information during a call Multimodal communications: Voice, Video, Chat, file sharing, over the same communications pipe
SIP Trunks offer the ability to have a voice call over
disparate physical links
SIP Trunk can be implemented over a wide variety of IP
communications trunks (i.e. Metro Ethernet, WiFi, GSM Cellular)
SIP Trunks offer the ability to have
improved redundancy
for communications
IP Links can be built with redundancy of communications methods and fast failover that results in quicker time to repair in case of failure
Agenda
Technology Overview
Deployment Scenarios
Issues with SIP Trunks for PSTN Access
What's in a SIP Trunk?
What services should customers look for in SIP Trunks?
What services should Service Providers offer with SIP Trunks?
Using the Cisco Unified Border Element (formerly Cisco Multiservice IP-to-IP Gateway) as a Border Element between Service Provider and Customer
Issues and configuration options for Cisco Unified Communications Manager Express (CUCME) with SIP Trunks for PSTN Access
Issus and configuration options for Cisco Unified Communications Manager (CUCM) with SIP Trunks for PSTN Access
General Troubleshooting Issues with SIP Trunks
Deployment
Scenarios
Deployment Scenarios
Basically, there are three methods of deploying SIP Trunks today, Centralized where trunks for all regions are centralized and provided only from a central
location, distributed, where each regional office has SIP Trunk from the providers and Hybrid models where different solutions are provided for different types
of traffic
Centralized Location of All SIP Trunks
Distributed Trunks to All Locations Hybrid Trunk Deployment, Deploy Trunks Based on Function
All PSTN Trunks are
removed from remote sites and replaced with SIP Trunks that terminate only at the central datacenter or
headquarters. This HQ site receives and routes ALL PSTN traffic via SIP Trunks to Service Provider.
SIP Trunks are provided from the Service Provider to all sites. Each site removes their PSTN access and instead replaces it with SIP Trunks from the
provider that terminate at the remote sites. Provider needs to route phone calls to remote site via SIP trunk at remote sites.
SIP Trunks are added in Headquarters and /or remote sites to complement PSTN trunks. Dialplan is altered so that traffic can flow across most effective trunk, and traffic can be effectively routed via both HQ and remote site Trunks.
Centralized Deployment Model
Distributed Deployment Model
Hybrid Deployment Model
Calls routed via
Centralized SIP Trunk
Calls Routed via PSTN
Calls Routed via Remote site SIP Trunk
Agenda
Technology Overview
Deployment Scenarios
Issues with SIP Trunks for PSTN Access
What's in a SIP Trunk?
What services should customers look for in SIP Trunks?
What services should Service Providers offer with SIP Trunks?
Using the Cisco Unified Border Element (formerly Cisco Multiservice IP-to-IP Gateway) as a Border Element between Service Provider and Customer
Issues and configuration options for Cisco Unified Communications Manager Express (CUCME) with SIP Trunks for PSTN Access
Issus and configuration options for Cisco Unified Communications Manager (CUCM) with SIP Trunks for PSTN Access
General Troubleshooting Issues with SIP Trunks
Issues with SIP Trunks for PSTN Access
Interoperability with IP PBX
Voice Band Data
Fax Call
Supplementary Features
Interoperability Issues with SIP Trunks
There is currently no standard for SIP Trunks that can provide the same level of consistency and interoperability of PSTN ISDN
Trunks
There are efforts underway in the industry to have more
interoperability; various efforts are being lead by the SIP forum, ATIS, TISPAN
The problem of interoperability is reduced by having a customer owned border element that can provide signaling interworking and transcoding
This problem can be further reduced by having a Service Provider owned Border Element on Customer premise that acts as a
demarcation point for signaling
Customer should test / test / test before deployment or first SIP Trunks solution, but replicate after that for scaling
Voice Band Data
Voice Band Data (VBD) is the ability to send information such as slow speed modem calls for credit card transactions or alarm systems (i.e. Telematics) across the voice channel of an PSTN connection
Voice Band Data can work reliable up to 56K with PSTN connections
With any codecs you cannot maintain a PCM clock sync so 56K connections are not possible; but medium speed modem connections are possible over G711
With compressed codecs (i.e. G729), you cannot reliable send modem tones over VoIP calls, so only low speed connections
VBD cannot be “guaranteed”, so an important consideration is whether there are PSTN circuits that can be left to support this at the site where SIP Trunks are being considered. The three most used types of VBD are:
Baudot connections for deaf users Credit card validation systems Security systems
These systems should all be tested before a SIP Trunk for PSTN access is used as a replacement Sending a Modem Call Over a Codec Is Like Putting It Through a Cheese Grater:
the Signal Will Never Be the Same
Fax Calls
SIP Trunks can typically use three different methods to supports FAX calls
T.38 FAX capabilities are exchanged
All Calls are sent as G711 and best effort fax is done
Call sends a RE-INVITE to up-speed to G711 when a FAX tone is detected
SIP Service provides also occasionally offer a separate fax to email service using T.37 Store and Forward fax
Fax Method T.38 Fax Capabilities Exchanged as Part of SIP Messages
All Call Sent as G711 Fax Tone Is Detected and RE- INVITE to up-speed to G711 Is sent
Pros Highest fax success rates can be achieved
Cleanest solution from signaling and media point of view
Use less bandwidth than G711
Fax and Voice calls differentiated
Most widely deployed
Simplest solution
Provides benefits of least bandwidth with G729 call initially upspeeding to G711 if call is FAX
Tone (2100Hz) can be mixed between Modem and Fax
Fax Pass-Through Cons Degree of interoperability
Not offered by many Service Providers
Consumes a large amount of bandwidth for all calls
No ability to distinguish FAX calls from Voice calls in CDRs
Each vendors support of RE-INVITEs is different
Currently not supported with all Cisco equipment
Supplementary Services
The supplementary service invoked over the SIP Trunk is not supported or understood by the far end SIP switch
For example, the signaling to place a call on HOLD and temporarily stop media can be done in one of several ways, all of them are compliant with the standard. Mismatching methods may be supported between two SIP switches
Testing of Supplementary Services before deployment is only way to ensure success
Create a testcase for each service before deployment Report findings to Service Provider
Determine if lack of these functionality should effect deployment
Typical Supplementary Services test cases
Placing call on HOLD
Forward on Busy/No Answer to Number within premise Transferring call to another extension
Correct billing for forwarded calls
PSTN
All Signaling Is Translated Resulting in
Fewer Interop Issues
SIP Signaling End-to-End Causes Interop Issues
SIP Network
Quality Control
As customers have deployed SIP Trunks for PSTN access, the
experience for users has been “inconsistent” (i.e. one calls is great, next is not great)
A “best practice” is to create a method of flagging calls (either via CDRs analysis or user feedback) that are very bad
Use data from CDRs (i.e. Jitter, Packet Loss) to determine if there are trends and average; these statistics can be gathered from the Customer premise Border Element
Try to determine if quality issues co-relate with specific events, such as dialing to some area codes or countries or specific times of day; service providers have different methods of routing that can effect quality
Service providers should ensure that they have a method of
measuring quality all the way to the customer premise; this can be used to distinguish their service from others
Agenda
Technology Overview
Deployment Scenarios
Issues with SIP Trunks for PSTN Access
What's in a SIP Trunk?
What services should customers look for in SIP Trunks?
What services should Service Providers offer with SIP Trunks?
Using the Cisco Unified Border Element (formerly Cisco Multiservice IP-to-IP Gateway) as a Border Element between Service Provider and Customer
Issues and configuration options for Cisco Unified Communications Manager Express (CUCME) with SIP Trunks for PSTN Access
Issus and configuration options for Cisco Unified Communications Manager (CUCM) with SIP Trunks for PSTN Access
General Troubleshooting Issues with SIP Trunks
SIP Trunk for PSTN—Service Offerings
Requirements Unacceptable Offering
Good Offering Best Offering
Security None IP Address Validation of SIP INVITES
TLS Signed SIP INVITES
Fax None G711 for all Calls T.38 support
Voice Band Data None Offer to work each issue individually
Offers SLA for Data speeds for VDB over SIP Trunks
Uptime None SLA with 95% uptime offered SLA with 99.999% uptime offered and access from customer for reports with refunds for nonconformance
Calling Plans N/A Per minute Flat rate with no cost calls between customers; each trunk can configure their own calling plan; billing records provided via WEB interface
Redundancy None Ability to route calls to a different phone number or IP address when trunk is down
Call re-routed in real time when the SIP Trunk fails; routing is to both secondary IP address and PSTN number
Call Number Porting
None Call porting of phone numbers can be
accomplished for some area codes within 30 days
All area codes can have phone number ported with zero lost calls in 48 hours
Cisco Unified
Border Element
(Formerly Cisco
Multiservice IP-to-
IP Gateway) as a
Border Element
Cisco Unified Border Element as a Border Element for SIP Trunk Solutions
Cisco Unified Border Element
(formerly Cisco Multi-service IP to IP Gateway)
Session
Management
Inter-working
Security
Demarcation
H323 to SIP SIP to SIPSIP Profiles & Variants
IOS Firewall Integration RTP Media Validation Signaling Protection Call Admission Control IP QOS/SLA Fault Isolation
Call Accounting
Topology Hiding
Cisco Unified
Border
Element
Cisco Unified
Border
Element
Cisco UBE allows the network to provide these services
Co-existence with other services such as MTP, SRST, TDM GW
CUBE Features
Network/Topology Hiding for Voice and Video Calls
Protocol Support—H.323 and SIP
Voice Codecs—G.711, G.729, G.726, G.723, G.728, Transparent
Video Codecs—H.261, H.263 and H.264
Codec Filtering
Media—Media Flow Through and Media Flow Around
DTMF Interworking—H.245 Alphanumeric, Signal, RFC2833, SIP NOTIFY
Fax/Modem—T.38, Passthrough, Cisco Fax Relay, Modem Passthrough
Security—TLS, IPSec with SRTP
Signaling Interworking
Supplementary Services
Transcoding
Transport Mode—TCP, UDP
Number Translation
Quality of Service
Call Admission Control
Call Detail Records
TCL/VXML Support Rotary Support H.323/SIP Trunk Cisco Unified Border Element SIP Trunk Cisco Unified Communications Manager Cluster Service Provider SBC
Connect Method Feature Over the Top Managed Router Managed Router Running NAT/PAT Managed Router Running Cisco UBE (IP2IP GW)
Voice Calls Possible X X X X
QoS Can Be Guaranteed X X X
Security X X X
IP Address Hiding X X
Call Counting X
Signaling Interworking (H323/SIP) X DTMF Interworking (Inband to OOB) X Transcoding (Any to Any Codec, etc.) X TCL/VxML (Ability to Run Scripts on Calls) X
Redundancy (HSRP) X X
Simple Interconnect SP Configuration with Multiple Endpoints X Per Call Voice Quality Statistics X CDR Collection Point for Multiple Entities X Support for REFER (Note: NOTIFY on DEMAND, Not Subscribed) X X Support for REFER with NSS to Pass Information X X
Media Flow-Through vs. Flow-Around
Media Bypasses the CUBE Gateway Media Flow-Around
Signal Leg: 1
Media Leg: 1
Media Leg: 2
Media Flow-Through
SBC
Signal Leg: 2
Signal Leg: 1
Signal Leg: 2
Address Hiding
Cisco UBE can hide the customer’s IP addresses by presenting its own IP address to the public side
Cisco UBE can also provide a solution for a customer’s multisite deployment in which there are overlapping IP addresses
Cisco UBE acts like a Back-to-Back User Agent—it would reformulate a request with entirely new From, Via, Contact, Call-ID, etc.
RTP headers are changed when configured for media flow-through
CUCM Cluster Site A CUBE Gateway Site A—192.168.10.x/24 192.168.10.10 192.168.10.50 Third Party Application Server 151.10.10.1 151.10.10.2 IP WAN 151.10.10.0/27 SBC
In Leg Out Leg Support
Fast Start Fast Start Bi-Directional Slow Start Slow Start Bi-Directional Fast Start Slow Start Bi-Directional
H.323-H.323
In Leg Out Leg Support
Early Offer Early Offer Bi-Directional Delayed Offer Delayed Offer Bi-Directional Delayed Offer Early Offer Uni-Directional
H.323-SIP
SIP-SIP
In Leg Out Leg Support
Fast Start Early Offer Bi-Directional Slow Start Delayed Offer Bi-Directional
Protocol Interworking
12.5( 1st)T
Delayed Offer (DO) – Early Offer (EO)
Provide a mechanism to translate a SIP DO to EO
Uni-directional only (EO to DO is not required/needed) Designed to enable Service Provider interconnects
Allows SP SIP Trunks (that often support only EO) to interconnect to CUCM 5.x/6.x SIP Trunks where DO is preffered
CUCM 5.x/6.x supports EO for outbound audio calls only with a G711 MTP - This DO to EO feature removes the MTP requirement
CUCM 5.x/6.x doesn’t support EO for video calls
Release:
Audio – 12.5(1st)T, Video – 12.5(2nd)T
12.5( 1st)T
DTMF Interworking
RFC2833 RFC2833 SIP SIP RFC2833 RFC2833 RFC2833 H.245-Signal RFC2833 H.245-Alphanumeric SIP H.323 G711 In Band Voice RFC2833 RFC2833 G711 In Band VoiceMedia Transcoding
CUBE supports universal
transcoding any voice codec to any other codec
Up to 400 sessions on 3845 Re-packetization with different sample sizes not supported
SP VoIP Network Enterprise Network iLBC, iSAC, Speex Internet Telephony IP Phones: G.711, G.729, G.722
Supported Codecs Release
G.711 a-law 64Kbps 12.4(11)XW G.711 µlaw 64Kbps 12.4(11)XW G.723 – 5.3 & 6.3 Kbps 12.4(11)XW G.729 (all variants) 8Kbps 12.4(11)XW iLBC – 13.3 & 15.2 Kbps 12.4(11)XW G.722 – 64kbps 12.5(1st)T CUBE CUBE
CUBE Transcoding: G.711, G.723.1, G.726, G.728, G.729/a, iLBC, G.722
12.5( 1st)T
IOS Firewall Support – SIP ALG
SIP support added to ALG, allowing FW inspection of:
Support for RFC3261
OPTIONS, INVITE, REGISTER, ACK, CANCEL, BYE Support for RFC3261 extension methods
INFO (RFC2976), PRACK (RFC3262), SUBSCRIBE/NOTIFY (RFC3265), UPDATE (RFC 3311), REFER
(RFC3515/RFC3892), MESSAGE (RFC3428)
Protocol conformance check: Enforcement of mandatory and forbidden header fields
SIP Dialog and transaction enforcement SIP over TCP
Media negotiation (RFC3264) Early and Late Media (RFC3960)
12.5( 1st)T
IOS FW Support – SIP Application Inspection
and Control
Filter out a black/whitelist of Callers or Callees
Filter SIP messages based on SIP Methods, SIP
header fields, or content-type
12.5( 1st)T
! Match SIP methods
match request method <method> ! Match header fields
match {request | response} header <header-name> regex <regex-param-map> ! Match SIP Status line
match response status regex <regex-param-map>
Validation of max-forwards Disable IM
Example: Block SIP messages coming from a particular proxy
parameter-map type regex unsecure_proxy pattern “compromised.server.com”
class-map type inspect sip sip_class
match request header Via regex unsecure_proxy policy-map type inspect sip sip_policy
class type inspect sip sip_class reset log
IOS FW Support – AIC Rate-Limiting
Rate-limits application messages Protects against DOS attacks
Uses the rich match criteria for AIC classes
Example: Limit to 10 SIP INVITE messages per second
12.5( 1st)T
class-map type inspect sip my_sip_class match request method invite
policy-map type inspect sip my_sip_policy class type inspect sip my_sip_class
rate-limit 10
class-map type inspect sip match-all my_sip_class match request method invite
match request header length gt 1026 policy-map type inspect sip my_sip_policy
class type inspect sip my_sip_class rate-limit 16
Example: Limit to 16 SIP INVITE requests whose header length is greater than 1026
Call Admission Control
1.
RSVP (only if IP-to-IP Gateway is used on both ends)
2.
Total calls
3.
CPU
4.
Memory
5.
IP call capacity
6.
Max-connections
SIP Trunks for
Cisco Unified
Communications
Manager Express
(CUCME)
Cisco Unified Communications Manager
Express
Supported on 3.4 on 12.4(4)T1
Supported on 4.0 on 12.4(9)T1
Configuration Options
for Cisco Unified
Communications
Manager (CUCM)
with SIP Trunks for
PSTN Access
SCCP MGCP H.323 CTI SIP/SIMPLE/KPML Cisco Unified Communicator
Cisco Unity®/Cisco
Unity Connection
CTI Apps Gateways
MeetingPlace/
MP Express Cisco Unified
Presence Server Carriers/ Other PBXs Cisco and 3rd-Party Phones Soft Phones Video Endpoints CUCME Microsoft LCS IBM Sametime
Cisco Unified Communications Manager
Release 5.0 - SIP Trunks
Cisco Communications Manager 5.0 integrates rich, native SIP and
SIMPLE support on both line-side and trunk-side interfaces (for both audio and video calls) with integrated presence on phones and applications; KPML and RFC 2833 support for DTMF; TLS and Digest Authentication for security; seamless protocol inter-working between SIP, H.323, MGCP,
Cisco
Communications Manager 5.0
Cisco Communications Manager 5.0
Cisco Communications Manager Trunks-
SIP Media Support
Cisco Communications Manager
supports receiving Early Offer and
Delayed Offer
Cisco Communications Manager sends
Delayed Offer to the callee—unless “MTP required” is checked—then Early Offer is used
Support for Delayed Offer is mandatory in RFC 3261:
“Concretely, the above rules specify two exchanges for UAs compliant to this specification alone—the offer is in the INVITE, and the answer in the 2xx (and possibly in a 1xx as well, with the same value), or the offer is in the 2xx, and the answer is in the ACK. All user agents that support INVITE must
support these two exchanges.“
Cisco Communications Manager SIP
Trunks Using Early Offer
SIP Early Offer—Outbound Calls must use an MTP
If no MTP resources available, call reverts to Delayed Offer Asymmetric EO DO is supported
MTPs are available in three forms:
Software based MTPs in Cisco IOS®-based gateways (available with any Cisco
IOS T-train software and scaling up to 500 sessions (calls) on the Cisco 3845 router platform)
Hardware based MTPs in Cisco IOS-based gateways (available with any Cisco IOS T-train software release hardware MTPs use on board DSP resources and scale calls according to the number of DSPs supported on the Cisco router platform)
Software based MTPs using the Cisco Communications Manager IP Voice Streaming application on an Cisco MCS server
MTPs (and Transcoders) can be controlled by Cisco
CUCM SIP Trunking—
SIP Delayed Offer, G711 & G729 Regions - No MTP
SIP Delayed Offer
G711 & G729 Regions between devices and SIP Trunk
For CUCM outbound calls—Codec Preference G711
For Inbound Calls to Cisco Communications Manager
SIP Delayed Offer calls—SIP switch selects codec SIP Early Offer calls—CUCM selects codec—G711
Fax calls—Fax Pass-through and T.38 Fax Relay
Inbound and Outbound Re-Invites supported
Voice Gateway SP SIP Switch
SIP Signaling 2.7XXX SIP RTP Media Stream 7776 5000 5001 5555 7777 RTP Media Stream CUBE
CUCM Designs—
SIP Delayed Offer, G711 and G729 Regions—No MTP
All Fax Machines are in G711 Region
All remote branch phones are in G729 Region
Voice calls, Fax pass-through, T.38 Fax Relay, G711 MOH For Outbound Calls—CUCM Codec Preference—region
dependent
For Inbound Calls—For G711 regions SP SIP switch Codec Preference
Central Site
Remote
Site G729 Inter Region Codec
G711 Inter Region Codec
CUBE
IP WAN
G729 SP WAN
CUCM SIP Trunking—Cube DO to EO
CUCM Delayed Offer, G711 & G729 Regions
Service Provider requires Early Offer
SIP Early Offer (meaning SDP included in INVITE) required by Service Provider
No MTP required in RTP path for Outbound CUCM calls
G711 & G729 Regions between devices and SIP Trunk
For CUCM outbound calls—SP selects codec
For Calls Inbound to Cisco Communications Manager
SIP Delayed Offer calls—SP selects codec SIP Early Offer calls—CUCM selects codec
Fax calls—Fax Pass-through and T.38 Fax Relay
Inbound and Outbound Re-Invites supported
Voice Gateway SP SIP Switch
SIP Delayed Offer 2.7XXX SIP RTP Media Stream 7776 5000 5001 5555 7777 RTP Media Stream CUBE MTP
CUCM SIP Trunking—Cube DO to EO
CUCM Delayed Offer, G711 & G729 Regions
Service Provider requires Early Offer
All Fax Machines are in G711 Region
All remote branch phones are in G729 Region
Voice calls, Fax pass-through, T.38 Fax Relay, G711 MOH
For Outbound Calls—SP selects codec
For Calls Inbound to Cisco Communications Manager
SIP Delayed Offer calls—SP selects codec SIP Early Offer calls—CUCM selects codec
Central Site
Remote
Site G729 Inter Region Codec
G711 Inter Region Codec
CUBE
IP WAN G729
SIP Switch
MTP G711 SP WAN
Current SIP Trunk Recommendations
and Strategy
Cisco Communications Manager 5.X preferred and
recommended over CM4.X implementations
Cisco Unified Border Element (CUBE) is recommended
as an Enterprise owned Border Element
SIP Delayed Offer is preferred over SIP Early Offer
Where SIP Early Offer is required by SP use CUBE DO
to EO to avoid MTP usage
CM7.0—Introduces:
G729 MTP for SIP Early Offer Support for + character
Implementation Options
CUCM to CUBE
CUCM CUBE redundancy Call Volume
CUBE Redundancy
CUBE Gateway to Service Provider
SIP Routing Entity
CUBE CUBE
CUCM - CUBE Redundancy
H.323 or SIP
Define each CUBE device by its IP address
SIP: Each CUBE GW is an IP address destination for SIP trunk H.323: Each CUBE GW is an H.323 Gateway
Define one route group is defined for each CUBE
Gateway;
then, define one route list to encompass all route
groups
Use TCP for fast failover on Trunk failure or tune UDP timers
CUBE Gateway 1 Route Pattern xxxx Route List Route Group N Route Group 1 CUBE Gateway N CUCM Cluster CUBE CUBE
CUBE
Call Volume
Add the number of CUBEs required
CUBE CUBE
SIP Proxy Server of Service
Performance Capacity Recommendations
for CUBE Gateway Platforms
Platform CUBE GW Only(VAD ON)
Multiple Features CUBE GW (VAD OFF) CUBE GW with Software MTP AS5000XM 1000 600 N/A 3845 750 525 250 3825 600 420 200 2851 600 280 125 2821 400 225 106 2811 200 112 53
CUBE Gateway Redundancy
HSRP
1:N redundancy
CUBE Broadsoft AS/NS MCI NS/RS CUBE CUBE CUBE HSRP 1:NCUBE Gateway to Service Provider—
Use DNS SRV
_sip._udp.cluster-ipipgw IN SRV 100 10 5060 ipipgw1.cisco.com IN SRV 100 10 5060 ipipgw2.cisco.com IN SRV 100 10 5060 ipipgw3.cisco.com ipipgw1 IN A 166.34.96.5 ipipgw2 IN A 166.34.96.6 Ipipgw3 IN A 166.34.96.7 _sip._udp.bsas IN SRV 100 50 5060 bsas1.vzb.com IN SRV 200 50 5060 bsas2.vzb.com bsas1 IN A 166.34.87.25 bsas2 IN A 166.34.87.26 INVITE [email protected] From: cluster_ipipgw.cisco.com FQDN: cluster_ipipgw.cisco.com FQDN: bsas.vzb.com DNS Server INVITE 2125551000@cluster- ipipgw.cisco.com From: bsas.vzb.com CUBE CUBE CUBEIncoming VoIP Call Outgoing VoIP Call
dial-peer voice 1 voip destination-pattern 1000 incoming called-number .T
session target ipv4:192.168.10.50 codec g711ulaw
dial-peer voice 2 voip destination-pattern 2000 session protocol sipv2
session target ipv4:10.10.10.5 codec g711ulaw
Basic CUBE Configuration
1. Enabling the IP-to-IP Calls
SBC3845#config t
SBC3845(config)# voice service voip
SBC3845(conf-voi-serv)#allow-connections h323 to h323
SBC3845(conf-voi-serv)#allow-connections h323 to sip
SBC3845(conf-voi-serv)#allow-connections sip to h323
SBC3845(conf-voi-serv)#allow-connections sip to sip
2. Mandatory to have Incoming and Outgoing VoIP Dial-peers
with required parameters like Protocol, Transport, Codec, CAC, QoS, etc.
1000 2000
Agenda
Technology Overview
Deployment Scenarios
Issues with SIP Trunks for PSTN Access
What's in a SIP Trunk?
What services should customers look for in SIP Trunks?
What services should Service Providers offer with SIP Trunks?
Using the Cisco Unified Border Element (formerly Cisco Multiservice IP-to-IP Gateway) as a Border Element between Service Provider and Customer
Issues and configuration options for Cisco Unified Communications Manager Express (CUCME) with SIP Trunks for PSTN Access
Issus and configuration options for Cisco Unified Communications Manager (CUCM) with SIP Trunks for PSTN Access
General Troubleshooting Issues with SIP Trunks
Cisco CUBE/SBC Portfolio
Platform Support
Scale Scale Scale Cisco 2600XM Cisco Cisco 2600XM 2600XM Cisco 2800 ISR Cisco 2800 Cisco 2800 ISR ISR Cisco 3700 Cisco 3700 Cisco 3700 Cisco 3800 ISR Cisco 3800 Cisco 3800 ISR ISR Cisco 7200 VXR Cisco 7200 Cisco 7200 VXR VXR Cisco 7301 Cisco 7301 Cisco 7301 AS5000XM AS5000XM AS5000XMCisco CUBE supports
1. SIP/H.323 2. CUCM interworking 3. Demarcation 4. Security 5. Transcoding Cisco XR12000 Cisco XR12000 Cisco XR12000 Cisco 7600 Cisco 7600 Cisco 7600 Cisco SBC supports
1. NAT & FW traversal
2. Security
3. CAC & Policies
4. Media & Protocol interworking
CUBE Licensing
FL-CUBE-25
FL-CUBE-100
Licenses are
additive,
increments of 25
sessions
Orderable on IOS
images of IP
Voice and up
FL-CUBE-25 USD $2900Cisco Unified Border Element License for up to 25 Sessions
1
FL-CUBE-100 USD $9900 Cisco Unified Border Element License for up to 100 Sessions
1
New Feature Licenses will be available to order
Cisco Unified Border Element
Enhanced Functionality with IOS Image
H323 – H323 Voice Calls
SIP TLS Secure RTP
Video Call Flow
H323 to SIP
SIP – SIP Voice Calls
SIP ALG and FIREWALL
Fe atu re s En ha nc ed w ith IO S Ima ge
Flow Around Media H.323 Gatekeeper -IVS- Image Security Images IP Voice
FL-CUBE License will be available as an option on several different IOS images. More powerful/expensive IOS images provide
Additional Resources
Cisco Multiservice IP-to-IP Gateway
General Information (Datasheet, Q and A)
http://www.cisco.com/en/US/products/sw/voicesw/ps5640/index .html
Cisco Multiservice IP-to-IP Gateway
Configuration Guide
http://www.cisco.com/en/US/products/sw/voicesw/ps5640/produ cts_installation_and_configuration_guides_list.html
Recommended Reading
Continue your Networkers at
Cisco Live learning experience with further reading from Cisco Press® Suggested books:
Cisco Voice Gateways and Gatekeepers [1-58705-258-X]
Cisco IP Communications Express:
Cisco Communications Manager Express with Cisco Unity Express [1-58705-180-X]
