LTRCOL-2310
Deploying SIP Trunks with Cisco
Unified Border Element (CUBE)
Enterprise and CUCM
Hussain Ali, CCIE# 38068 (Voice, Collaboration)
Technical Marketing Engineer
Dilip Singh, CCIE# 16545 (Collaboration)
Technical Leader
Objectives
•
Provide a quick overview of SIP Trunking with CUBE
•
Understand and deploy a working ITSP SIP trunk for making and receiving
calls
•
Understand how to capture and analyze CUBE debugs to troubleshoot SIP
issues using available tools
•
To leave participants with good understanding of CUCM, CUBE and
MediaSense SIP Trunk operation and monitoring
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
3
Cisco Webex Teams
Questions?
Use Cisco Webex Teams (formerly Cisco Spark)
to chat with the speaker after the session
How
1 Find this session in the Cisco Events Mobile App
2 Click “Join the Discussion”
3 Install Webex Teams or go directly to the team space
4 Enter messages/questions in the team space
cs.co/ciscolivebot#LTRCOL-2310
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
4
CUBE Overview
Enterprise
LAN
Unified CM
DEMARC
Collaboration Deployment
ITSP
WAN (SIP Provider)
PSTN (PRI/FXO)
TDM Backup
(Not available in
vCUBE)
Gig0/0
PSTN
Gig0/1
CUBE
DEMARC
SIP
H.323
RTP
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
6
CUBE (Enterprise) Product Portfolio
50-150
ASR 1004/6 RP2
Introducing CUBE on
CSR
vCUBE [Performance
50-100
ASR 1002-X
ASR 1001-X
ISR 4451-X
dependent on vCPU and
memory]
CPS
20-35
ISR 4431
3900E Series ISR-G2
(3925E, 3945E)
ISR 4351
17
3900 Series ISR-G2 (3925, 3945)
ISR-4K (4321, 4331)
8-12
Note: SM-X-PVDM module
supported on XE3.16 or later
for ISR 4K platforms
2900 Series
ISR-G2 (2901, 2911, 2921, 2951)
<5
800 ISR
4
<50
500-600
900-1000
2000-2500
4000
4500-6000
7000-10,000
12K-14K
14-16K
Active Concurrent Voice Calls Capacity
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
7
CUBE Software Release Mapping
ISR G2
ASR 1K / ISR-4K/vCUBE (CSR)
CUBE
Vers.
2900/
3900
FCS
CUBE
Vers.
IOS XE Release 16 2
FCS
11.5.14
15.6(2)T14
Mar 2016
N/A 3
16.2.13
Mar 2016
11.5.2
15.6(3)M1
Dec 2016
11.5.23
16.3.2/16.4.13
Nov 2016
EOL
EOL
EOL
11.6.0
16.5.1
Mar 2017
EOL
EOL
EOL
12.0.0
16.6.1
July 2017
EOL
EOL
EOL
12.0.0
16.7.1
Nov 2017
EOL
EOL
EOL
12.1.0
16.8.1
Mar 2018
EOL
EOL
EOL
12.2.0
16.9.1
July 2018
EOL
EOL
EOL
12.5.0
16.10.1a
Nov 2018
2 IOS-XE 16 requires a minimum of ASR1001-X, 1002-X, 1004/1006 RP2, ESP20 (Embedded Service Processor, SIP40 (SPA Interface processor)
3 IOS-XE release 16.2.1 does not support CUBE functionality on the platforms. There is no CUBE version 11.5.1 for the XE based pla tforms. All CUBE features from 11.5.0 (IOS-XE 3.17) and earlier versions
along with CUBE 11.5.1 (March 2016 release) on ISR G2 are included in CUBE release 11.5.2 for the IOS -XE based platforms, IOS-XE release 16.3.1 [July 2016 release]
4 IOS 15.6(2)T will show CUBE Release version to be 12.0.0 but due to DDTS# CSCuz43735, rebuilds for this release train will align to CUBE release 11.5.1, that is 15.6(2)T1/T2/T3/T4 and so on will be
CUBE version 11.5.1
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
8
CUBE Call Flow
CUBE Call Processing
•
Actively involved in the call treatment, signalling
and media streams
•
•
•
SIP B2B User Agent
Provides full inspection of signalling, and protection
against malformed and malicious packets
Media Flow-Through
•
Media Flow-Around
 Signaling and media terminated by the Cisco
Unified Border Element
 Transcoding and complete IP address hiding
require this model
CUBE
Media is handled in two different modes:
•
IP
Media Flow-Through
Signalling is terminated, interpreted and reoriginated
•
•
CUBE
IP
Media Flow-Around
Digital Signal Processors (DSPs) are only required
for transcoding (calls with dissimilar codecs)
LTRCOL-2310
 Only Signaling is terminated on CUBE
 Media bypasses the Cisco Unified Border
Element
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
10
Cisco Unified Border Element Basic Call Flow
Originating
Endpoint 1000
voice service voip
mode border-element
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
Incoming VoIP Call
Outgoing VoIP Call
Terminating
Endpoint –
2000
CUBE
dial-peer voice 1 voip
destination-pattern 1000
session protocol sipv2
session target ipv4:1.1.1.1
codec g711ulaw
dial-peer voice 2 voip
destination-pattern 2000
session protocol sipv2
session target ipv4:2.2.2.2
codec g711ulaw
1.
Incoming VoIP setup message from originating endpoint
2.
This matches inbound VoIP dial peer 1 for characteristics such as codec, VAD,
DTMF method, protocol, etc.
3.
Match the called number to outbound VoIP dial peer 2
4.
Outgoing VoIP setup message
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
11
Understanding the Call flow
Incoming VoIP Call Leg
Matches an Incoming Dial-peer
1000
1.1.1.1
VRF1 – 10.10.10.10
Outgoing VoIP Call Leg
Matches an Outbound Dial-peer
CUBE
20.20.20.20 – VRF2
INVITE /w SDP
INVITE /w SDP
c= 1.1.1.1
m=audio abc RTP/AVP 0
2000
2.2.2.2
c= 20.20.20.20
m=audio xxx RTP/AVP 0
100 TRYING
100 TRYING
180 RINGING
180 RINGING
200 OK
200 OK
c= 2.2.2.2
m=audio uvw RTP/AVP 0
c= 10.10.10.10
m=audio xyz RTP/AVP 0
ACK
ACK
1.1.1.1
BYE
200 OK
RTP (Audio)
10.10.10.10
20.20.20.20
2.2.2.2
BYE
200 OK
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
12
Basic Show Commands for Active Calls
CUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF2
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :
No. CallId
dstCallId LocalRTP
1
17
18
17474
2
18
17
17476
RmtRTP
6000
6001
LocalIP
10.10.10.10
20.20.20.20
RemoteIP
1.1.1.1
2.2.2.2
MPSS
NO
NO
VRF
VRF1
VRF2
Found 2 active RTP connections
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
13
Basic Show Commands for Active Calls
CUBE# show call active voice brief
121A
: 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF1
121A
: 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF2
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :
No. CallId
dstCallId LocalRTP
1
17
18
17474
2
18
17
17476
RmtRTP
6000
6001
LocalIP
10.10.10.10
20.20.20.20
RemoteIP
1.1.1.1
2.2.2.2
MPSS
NO
NO
VRF
VRF1
VRF2
Found 2 active RTP connections
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
14
Basic Show Commands for Active Calls
CUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF2
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :
No. CallId
dstCallId LocalRTP
1
17
18
17474
2
18
17
17476
RmtRTP
6000
6001
LocalIP
10.10.10.10
20.20.20.20
RemoteIP
1.1.1.1
2.2.2.2
MPSS
NO
NO
VRF
VRF1
VRF2
Found 2 active RTP connections
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
15
Basic Show Commands for Active Calls
CUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF2
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :
No. CallId
dstCallId LocalRTP
1
17
18
17474
2
18
17
17476
RmtRTP
6000
6001
LocalIP
10.10.10.10
20.20.20.20
RemoteIP
1.1.1.1
2.2.2.2
MPSS
NO
NO
VRF
VRF1
VRF2
Found 2 active RTP connections
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
16
Basic Show Commands for Active Calls
CUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF2
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :
No. CallId
dstCallId LocalRTP
1
17
18
17474
2
18
17
17476
RmtRTP
6000
6001
LocalIP
10.10.10.10
20.20.20.20
RemoteIP
1.1.1.1
2.2.2.2
MPSS
NO
NO
VRF
VRF1
VRF2
Found 2 active RTP connections
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
17
Basic Show Commands for Active Calls
CUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF2
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :
No. CallId
dstCallId LocalRTP
1
17
18
17474
2
18
17
17476
RmtRTP
6000
6001
LocalIP
10.10.10.10
20.20.20.20
RemoteIP
1.1.1.1
2.2.2.2
MPSS
NO
NO
VRF
VRF1
VRF2
Found 2 active RTP connections
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
18
Basic Show Commands for Active Calls
CUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF2
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :
No. CallId
dstCallId LocalRTP
1
17
18
17474
2
18
17
17476
RmtRTP
6000
6001
LocalIP
10.10.10.10
20.20.20.20
RemoteIP
1.1.1.1
2.2.2.2
MPSS
NO
NO
VRF
VRF1
VRF2
Found 2 active RTP connections
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
19
Basic Show Commands for Active Calls
CUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF2
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :
No. CallId
dstCallId LocalRTP
1
17
18
17474
2
18
17
17476
RmtRTP
6000
6001
LocalIP
10.10.10.10
20.20.20.20
RemoteIP
1.1.1.1
2.2.2.2
MPSS
NO
NO
VRF
VRF1
VRF2
Found 2 active RTP connections
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
20
Basic Show Commands for Active Calls
CUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF2
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :
No. CallId
dstCallId LocalRTP
1
17
18
17474
2
18
17
17476
RmtRTP
6000
6001
LocalIP
10.10.10.10
20.20.20.20
RemoteIP
1.1.1.1
2.2.2.2
MPSS
NO
NO
VRF
VRF1
VRF2
Found 2 active RTP connections
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
21
Basic Show Commands for Active Calls
CUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF2
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :
No. CallId
dstCallId LocalRTP
1
17
18
17474
2
18
17
17476
RmtRTP
6000
6001
LocalIP
10.10.10.10
20.20.20.20
RemoteIP
1.1.1.1
2.2.2.2
MPSS
NO
NO
VRF
VRF1
VRF2
Found 2 active RTP connections
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
22
Basic Show Commands for Active Calls
CUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF2
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :
No. CallId
dstCallId LocalRTP
1
17
18
17474
2
18
17
17476
RmtRTP
6000
6001
LocalIP
10.10.10.10
20.20.20.20
RemoteIP
1.1.1.1
2.2.2.2
MPSS
NO
NO
VRF
VRF1
VRF2
Found 2 active RTP connections
LTRCOL-2310
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23
Basic Show Commands for Active Calls
CUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF2
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :
No. CallId
dstCallId LocalRTP
1
17
18
17474
2
18
17
17476
RmtRTP
6000
6001
LocalIP
10.10.10.10
20.20.20.20
RemoteIP
1.1.1.1
2.2.2.2
MPSS
NO
NO
VRF
VRF1
VRF2
Found 2 active RTP connections
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
24
Basic Show Commands for Active Calls
CUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF2
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
CUBE# show voip rtp connections
VoIP RTP active connections :
No. CallId
dstCallId LocalRTP
1
17
18
17474
2
18
17
17476
RmtRTP
6000
6001
LocalIP
10.10.10.10
20.20.20.20
RemoteIP
1.1.1.1
2.2.2.2
MPSS
NO
NO
VRF
VRF1
VRF2
Found 2 active RTP connections
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
25
Basic Show Commands for Active Calls
CUBE# show call active voice brief
121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:1 Answer 1000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF1
121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:2 Originate 2000 active
dur 00:00:14 tx:0/0 rx:0/0
IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
VRF:VRF2
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
CUBE#
show voip rtp connections
VoIP RTP active connections :
No. CallId
dstCallId LocalRTP
1
17
18
17474
2
18
17
17476
RmtRTP
6000
6001
LocalIP
10.10.10.10
20.20.20.20
RemoteIP
1.1.1.1
2.2.2.2
MPSS
NO
NO
VRF
VRF1
VRF2
Found 2 active RTP connections
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
Transitioning to SIP
Trunking
Step 1: Configure CUCM to route calls
to the edge SBC
Module 1
SIP Trunk Pointing to CUBE
Standby
A
CUBE
Active
IP PSTN
CUBE
Enterprise
Campus
CUBE with High
Availability
MPLS
• Configure CUCM to route all PSTN
PSTN is now
calls (central and branch) to used
CUBE
only for
emergency
(Gig0/0
in our slides) via a SIP
trunk
SRST
calls over
FXO lines
• Make sure all different patterns of
CME
calls – local,
long distance,
international, emergency,TDM PBX
Enterprise etc.. are pointing to
informational
Branch Offices
CUBE
LTRCOL-2310
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28
Step 2: Get details from SIP Trunk provider
Module 1
Item
SIP Trunk service provider requirement
Sample Response
1
SIP Trunk IP Address (Destination IP Address for INVITES)
10.1.40.11 or DNS
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
SBC’s external IP address that is required for the SP to
accept/authenticate calls (Source IP Address for INVITES)
10.1.40.POD
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
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
Step 3: Enable CUBE Application on Cisco routers
Module 1
1. Enable CUBE Application
voice service voip
mode border-element license capacity 20  License count entered here not enforced though this CLI is
allow-connections sip to sip
required to see “show cube” CLI output
 By default IOS/IOS-XE voice devices do not allow an incoming
VoIP leg to go out as VoIP
2. Configure any other global settings to meet SP’s requirements
voice service voip
media bulk-stats  To increment Rx/Tx counters on IOS-XE based platforms. W/O this CLI, it will show 0/0
sip
early-offer forced
header-passing
error-passthru
3. Create a trusted list of IP addresses to prevent toll-fraud
voice service voip
ip address trusted list

ipv4 10.1.40.11 ! ITSP SIP Trunk
ipv4 198.18.133.3 ! CUCM
sip
silent-discard untrusted 
Applications initiating signalling towards CUBE, e.g. CUCM, CVP,
Service Provider’s SBC. IP Addresses from dial-peers with “session target
ip” or Server Group are trusted by default and need not be populated here
Default configuration starting XE 3.10.1 /15.3(3)M1 to mitigate TDoS Attack
LTRCOL-2310
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30
Step 4: Configure Call routing on CUBE
Module 1
• Dial-Peer – “static routing” table mapping phone numbers to interfaces or IP addresses
• LAN Dial-Peers – Dial-peers that are facing towards the IP PBX for sending and receiving
call legs to and from the PBX. Always bind LAN interface(s) on CUBE to LAN dial-peers,
ensuring SIP/RTP is sourced from the intended LAN interfaces(s)
• WAN Dial-Peers – Dial-peers that are facing towards the SIP Trunk provider for sending
and receiving call legs to and from the ITSP. Always bind CUBE’s WAN interface(s) to WAN
dial-peer(s).
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
31
Understanding Dial-Peer Matching Techniques:
Module 1
LAN & WAN Dial-Peers
•
LAN Dial-Peers – Dial-peers that are facing towards the IP PBX for sending and
receiving calls to & from the PBX. Should be bound to the LAN interface(s) of CUBE
to ensure SIP/RTP is sourced from the LAN IP(s) of the CUBE.
•
WAN Dial-Peers – Dial-peers that are facing towards the SIP Trunk provider for
sending & receiving calls to & from the provider. Should be bound to WAN
interface(s) of CUBE.
Inbound LAN Dial-Peer
A
Outbound Calls
CUCM SIP Trunk
Outbound WAN Dial-Peer
ITSP SIP Trunk
IP PSTN
CUBE
Inbound Calls
Inbound WAN Dial-Peer
Outbound LAN Dial-Peer
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
32
WAN Dial-Peer Configuration
Module 1
Inbound Dial-Peer for call legs from SP to CUBE
dial-peer voice 200 voip
description *** Inbound WAN side dial-peer ***
incoming called-number 408944….$
session protocol sipv2
voice-class sip bind control source gig0/1
voice-class sip bind media source gig0/1
codec g711ulaw
dtmf-relay rtp-nte
no vad
Specific to your DID range
assigned by the SP
Apply bind to all dial-peers when
CUBE has multiple interfaces.
Gig0/1 faces SP.
Outbound Dial-Peer for call legs from CUBE to SP
dial-peer voice 201 voip
description *** Outbound WAN side dial-peer ***
translation-profile outgoing Digitstrip
destination-pattern 81[2-9]..[2-9]......$
session protocol sipv2
voice-class sip bind control source gig0/1
voice-class sip bind media source gig0/1
session target ipv4:<SIP_Trunk_IP_Address>
codec g711ulaw
dtmf-relay rtp-nte
no vad
Translation rule/profile to strip the
access code (9) before delivering
the call to the SP
Dial-peer for making long distance
calls to SP, based on NANP (North
American Numbering Plan)
Note: Separate outgoing DP to be created for Local, International,
Emergency, Informational calls etc.
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
Module 1
LAN Dial-Peer Configuration
Inbound Dial-Peer for call legs from CUCM to CUBE
dial-peer voice 100 voip
description *** Inbound LAN side dial-peer ***
incoming called-number 8T
session protocol sipv2
voice-class sip bind control source gig0/0
voice-class sip bind media source gig0/0
codec g711ulaw
dtmf-relay rtp-nte
no vad
CUCM sending 8 (access code) + All
digits dialed
Apply bind to all dial-peers when
CUBE has multiple interfaces. Gig0/0
faces CUCM.
Outbound Dial-Peer for call legs from CUBE to CUCM
dial-peer voice 101 voip
description *** Outbound LAN side dial-peer ***
destination-pattern +1408944….$
session protocol sipv2
voice-class sip bind control source gig0/0
voice-class sip bind media source gig0/0
session target ipv4:198.18.133.3
codec g711ulaw
dtmf-relay rtp-nte
no vad
SP will be sending 10 digits (NANP)
based on your DID that is being
delivered to CUCM
CUCM IP Address
Default codec is G729 if none is
specified
Note: If more than 1 CUCM cluster exists, you will have to create multiple such LAN dial-peers with “preference CLI” for CUCM redundancy/load balancing as the
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
traditional way to accommodate multiple trunks
dial-peer voice 201 voip
description *Outbound WAN dial-peer. From CUBE to SP*
destination-pattern 81[2-9]..[2-9]......$
session protocol sipv2
session target ipv4:10.1.40.11
session transport udp
voice-class sip bind control source-interface Gig0/1
voice-class sip bind media source-interface Gig0/1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 100 voip
description *Inbound LAN dial-peer. From CUCM to CUBE*
session protocol sipv2
incoming called-number 8T
voice-class sip bind control source-interface Gig0/0
voice-class sip bind media source-interface Gig0/0
dtmf-relay rtp-nte
codec g711ulaw
no vad
Inbound LAN Dial-Peer
A
Outbound Calls
CUCM SIP Trunk
G0/0
ITSP SIP Trunk
CUBE
G0/1
10.1.40.11
198.18.133.3
Outbound LAN Dial-Peer
Outbound WAN Dial-Peer
Inbound Calls
dial-peer voice 101 voip
description *Outbound LAN dial-peer. From CUBE to CUCM*
translation-profile outgoing CUBE_to_CUCM
destination-pattern +1408944....$
session protocol sipv2
session target ipv4:198.18.133.3
voice-class sip bind control source-interface Gig0/0
voice-class sip bind media source-interface Gig0/0
dtmf-relay rtp-nte
codec g711ulaw
no vad
Inbound WAN Dial-Peer
dial-peer voice 200 voip
description *Inbound WAN dial-peer. From Provider to CUBE*
session protocol sipv2
incoming uri via 200
voice-class sip bind control source-interface Gig0/1
voice-class sip bind media source-interface Gig0/1
dtmf-relay rtp-nte
codec g711ulaw
no vad
voice class uri 200 sip
© 2019 Cisco and/or its affiliates. All rights reserved.
host ipv4:10.1.40.11
LTRCOL-2310
Cisco Public
35
OPTIONS KeepAlive
CUBE SIP Trunk Monitoring with OOD Options message
A
ITSP SIP Trunk
CUCM SIP Trunk
ITSP
CUBE
OOD Options
200 OK
Dial-Peer 201 = ACTIVE
INVITE
200 OK
• Out-of-dialog OPTIONS
message sent to check the
status of the SIP Trunk
• The dial-peer is “busyout” if it
does not receive a response
within a configurable time period
INVITE
200 OK
OOD Options
• For an INVITE that matches a
“busyout” dial-peer, CUBE
sends “503 Service Unavailable”
Timeout – no
response
Dial-peer 201 = BUSYOUT
INVITE
OOD Options
• If there is a secondary dial-peer
configured, the call will be rerouted the secondary path
503 Service Unavailable
OOD Options
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
37
CUBE SIP Trunk Monitoring with OOD Options message
A
SP SIP Trunk
CUCM SIP Trunk
CUBE
SP
SIP
OOD Options
200 OK
INVITE
DP 100 =
ACTIVE
Three timers that can be configured:
• up-Interval: OPTIONS keepalive timer interval
for UP endpoint
• down-interval: OPTIONS keepalive timer
interval for DOWN endpoint
• retry: Retry count for OPTIONS keepalive
transmission
INVITE
200 OK
200 OK
OOD Options
Timeout – no
response
DP 100 = BUSYOUT
INVITE
dial-peer voice 100 voip
voice-class sip optionskeepalive up-interval 20 downinterval 20 retry 3
Warning:
• Each dial-peer that has options message
configured sends out a separate message.
• EEM Script can be used to busyout other dialpeers
OOD Options
503 Service Unavailable
OOD Options
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
38
OOD OPTIONS Ping Keepalive Enhancement
• Each dial-peer that has OPTIONS message
A
SP SIP Trunk
CUCM SIP Trunk
CUBE
OOD Options (DP 201)
200 OK
DP 201 : Session Target IPv4:10.1.40.11
INVITE
200 OK
INVITE (DP 100)
200 OK
OOD Options (DP 211)
200 OK
DP 211: Session Target IPv4:10.1.40.11
OOD Options (DP 400)
200 OK
DP 400: Session Target IPv4:1.1.1.1
SP SIP
configured sends out a separate message, even if
the session targets are same
• Network bandwidth and process runtime are
wasted in CUBE and remote targets to sustain
duplicate OOD OPTIONS Ping heartbeat keepalive
connection
• Consolidate SIP OOD Options Ping connections
by grouping SIP dial-peers with same OOD
Options Ping setup
• New CLI : “voice class sip-keepalive-
profile <tag>” is used to define OOD OPTIONS
Ping setup
• Consolidated SIP OOD Options Ping connection
will then be established with a target for multiple
SIP dial-peers with the same target and OOD
Options Ping profile setup
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
39
OOD OPTIONS Ping Keepalive Enhancement –
Configuration
voice class sip-options-keepalive 1
description UDP Options consolidation
down-interval 49
up-interval 180
Single OOD Option
retry 7
Ping Group applied
transport udp
to multiple dial-peers
with same session
targets
dial-peer voice 201 voip
destination-pattern 6666
session protocol sipv2
session target ipv4:10.104.45.253
voice-class sip options-keepalive profile 1
dial-peer voice 211 voip
destination-pattern 5555
session protocol sipv2
session target ipv4:10.104.45.253
voice-class sip options-keepalive profile 1
Sample Show command output
CUBE#sh voice class sip-options-keepalive 1
Voice class sip-options-keepalive: 1
AdminStat: Up
Description: UDP Options consolidation
Transport: udp
Sip Profiles: 0
Interval(seconds) Up: 180
Down: 49
Retry: 7
Peer Tag
Server Group
OOD SessID
OOD Stat
IfIndex
--------
------------
----------
--------
-------
201
4
Active
9
211
4
Active
10
OOD SessID: 4
OOD Stat: Active
Target: ipv4:10.104.45.253
Transport: udp
Sip Profiles: 0
•
With OOD Options Ping Keepalive group, an options ping keepalive connection is established on per remote target base as opposed an options ping
keepalive connection established per dial-peer basis. Up to 10,000 “voice class sip-options-keepalive <tag>” can be defined per system
•
Either legacy “sip options-keepalive” or the new “sip options-keepalive profile <tag>” can be configured on a dial-peer. Dial-peers with Destination Server
Group instead of Session Target IP must use Options Keepalive Profile and not the legacy CLI.
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
40
Module 2
SIP Normalization
SIP profiles is a mechanism to normalise or customise SIP at the
network border to provide interop between incompatible devices
SIP incompatibilities arise due to:
• A device rejecting an unknown header (value
or parameter) instead of ignoring it
• A device expecting an optional header
value/parameter or can be implemented in
multiple ways
Add user=phone for INVITEs
Incoming
INVITE
sip:5551000@sip.com:5060
SIP/2.0
Outgoing
CUBE
INVITE
sip:5551000@sip.com:5060
user=phone SIP/2.0
voice class sip-profiles 100
request INVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
request REINVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
• A device sending a value/parameter that
must be changed or suppressed
(“normalised”) before it leaves/enters the
enterprise to comply with policies
• Variations in the SIP standards of how to
achieve certain functions
• With CUBE 10.0.1 SIP Profiles
can be applied to inbound SIP
messages as well
Modify a “sip:” URI to a “tel:” URI in INVITEs
Incoming
INVITE
sip:2222000020@9.13.24.6:5060
SIP/2.0
Outgoing
CUBE
INVITE
tel:2222000020
SIP/2.0
voice class sip-profiles 100
request INVITE sip-header SIP-Req-URI modify "sip:(.*)@[^ ]+" "tel:\1"
request INVITE sip-header From modify "<sip:(.*)@.*>" "<tel:\1>"
request INVITE sip-header To modify "<sip:(.*)@.*>" "<tel:\1>"
More information at http://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border-element/118825-technote-sip-00.html
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
41
SIP Profile Support for Non-Standard Headers
Module 2

Introducing support for adding/copying/removing/modifying non-standard SIP
headers using SIP profiles

A new 'WORD' option has been added to the SIP Profiles CLI chain to allow
the user to configure any non-standard SIP Header
CUBE(config)#voice class sip-profiles 1
CUBE(config-class)#request INVITE sip-header ?
Accept-Contact SIP header Accept-Contact
…….
Via
SIP header Via
WORD
Any other SIP header name
WWW-Authenticate
SIP header WWW-Authenticate
The new “WORD”
option for specifying
unsupported headers
CUBE(config-class)#request INVITE sip-header WORD ?
ADD
addition of the header
COPY Copy a header
MODIFY
Modification of a header
REMOVE
Removal of a header
CUBE(config-class)#request INVITE sip-header WORD ADD “MyCustomHeader : Hussain Ali”
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
Debugging Made Easier
Module 2
Categorize Debugs based on Functionality
 Categorization based on
Functionality
1.
2.
3.
4.
5.
6.
7.
Audio/video/sdp/control
Configuration /sip-transport
CAC
DTMF/FAX/Line-side
Registration
Sdp - passthrough
Sip-profile/SRTP/transcoder
Router# debug ccsip feature < audio | cac |
config | control | dtmf | fax | line | misc |
misc-features | parse | registration | sdpnegotiation | sdp-passthrough | sip-profiles |
sip-transport | srtp | supplementary-services
| transcoder | video >
Example: enabling DTMF and audio debugs only with default log level is considered.
DTMF(32) debug code
CUBE#sh debugging
CCSIP SPI: SIP info debug tracing is enabled (filter is OFF)
CCSIP SPI: audio debugging for ccsip info is enabled (active)
CCSIP SPI: dtmf debugging for ccsip info is enabled (active)
Audio(2) debug code
May 21 17:54:53.377: //444/5FE632EB8479/SIP/Info/verbose/32/sipSPI_ipip_store_channel_info: dtmf negotiation
done, storing negotiated dtmf = 0,
May 21 17:54:53.377: //444/5FE632EB8479/SIP/Info/info/2/sipSPIUpdateCallEntry:
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
43
Audio Transcoding and Transrating
iLBC, iSAC,
Speex
Enterprise
VoIP
ITSP
Module 3
IP Phones:
G.711, G.729 20 ms,
G.722
CUBE
G.729 30 ms
•
•
Transcoding (12.4.20T)
• One voice codec to any other codec E.g. iLBC-G.711 or iLBCG.729
• CUCM 7.1.5 or later supports universal Transcoding
Transrating (15.0.1M)
• Different packetizations of the same codec
•
E.g. G.729 20ms to G.729 30ms
• Support for SIP-SIP calls
• No sRTP support with transrating
dial-peer voice 2 voip
codec g729r8 bytes 30 fixed-bytes
!Call volume (gain/loss) adjustment
dial-peer voice 2 voip
audio incoming level-adjustment x
audio outgoing level-adjustment y
LTRCOL-2310
• Transcoding: G.711, G.723.1, G.726, G.728,
G.729/a, iLBC, G.722
• Transrating: G.729 20ms ↔ 30ms (AT&T)
Supported Codecs
Packetization
(ms)
G.711 a-law 64 Kbps
10, 20, 30
G.711 µlaw 64 Kbps
10, 20, 30
G.723 5.3/6.3 Kbps
30, 60
G.729, G.729A, G.729B, 10, 20, 30, 40, 50,
G.729AB 8 Kbps
60
G.722—64 Kbps
10, 20, 30
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44
Configuration for LTI based Transcoding
Module 3
(ISR-G2/4K & ASR)
1. Enabling dspfarm services
under voice-card
voice-card 0/1
dspfarm ! Only ISR G2
dsp services dspfarm
2. dspfarm profile configuration
dspfarm profile 3 transcode
codec g711ulaw
codec g711alaw
codec g729abr8
codec g729ar8
codec ilbc
maximum sessions 100
associate application CUBE
Feature Notes:
• This uses Local Transcoding Interface to
communicate between CUBE and DSPs
• Also available on ISR-G2 starting IOS 15.2.3T
• Can only be used if CUBE invokes the DSP
for media services
• CUCM cannot invoke DSPs using this LTI
interface
LTRCOL-2310
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45
Module 4, 5
Call Admission Control at the edge...
CUBE provides various CAC mechanisms to safeguard your network from SIP based attacks and to enforce policies based on:
• Total calls
• Maximum connections per destination
• CPU & Memory
• Dial-peer or interface bandwidth
• Call spike detection
Total Calls,
CPU, Memory
High Water Mark
Low Water Mark
Call Spike
Detection
CUBE
call spike call-number [steps
number-of-steps size milliseconds]
call spike 10 steps 5 size 200
CUBE
call threshold global [total/mem/cpu] calls low xx high yy
call treatment on
Max Calls per
Destination
Call #1
Max Bandwidth
based
Call #3 Rejected
by CUBE
Call #1 – 80Kbps
Call #2 – 80 Kbps
Call #2
Call #3
Call #3
Rejected by
CUBE
If a call spike is detected,
reject calls
CUBE
dial-peer voice 1 voip
max-conn 2
Call #3 – 80 Kbps
CUBE
dial-peer voice 1 voip
max-bandwidth 160
LTRCOL-2310
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46
Module 6
Destination Dial-peer Group
Allows grouping of outbound dial-peers based on an incoming dial-peer, reducing
existing outbound dial-peer provisioning requirements
• Eliminates the need to configure extra outbound dial-peers that are sometimes
needed as workarounds to achieve desired call routing outcome
•
•
Multiple outbound dial-peers are saved under a new “voice class dpg <tag>”. The
new “destination dpg <tag>” command line of an inbound voip dial-peer can
be used to reference the new dpg (dial-peer group)
•
Once an incoming voip call is handled by an inbound voip dial-peer with an active
dpg, dial-peers of a dpg will then be used as outbound dial-peers for an incoming
call
•
The order of outgoing call setups will be the sorted list of dial-peers from a dpg, i.e,
the destination-patterns of the outgoing dial-peers is not relevant for selection
LTRCOL-2310
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47
Module 6
Destination Dial-peer Group Configuration
dial-peer voice 1001 voip
destination-pattern 2222
session protocol sipv2
session target ipv4:10.1.1.1
!
dial-peer voice 1002 voip
destination-pattern 3333
session protocol sipv2
session target ipv4:10.1.1.2
!
dial-peer voice 1003 voip
destination-pattern 4444
session protocol sipv2
session target ipv4:10.1.1.3
voice class dpg 10000
description Voice Class DPG for SJ
dial-peer 1001 preference 1
dial-peer 1002 preference 2
dial-peer 1003
!
dial-peer voice 100 voip
description Inbound DP
incoming called-number 1341
destination dpg 10000
1. Incoming Dial-peer is first
matched
2. Now the DPG associated with
the INBOUND DP is selected
LTRCOL-2310
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48
Module 7
SIP Profile Tagging
• For tagging the rules:
voice class sip-profiles 1
rule 1 request INVITE sip-header Contact Modify “(.*)” “\1;temp=xyz”
rule 2 request INVITE sip-header Supported Add “Supported: ”
• For inserting a rule between two rules using “before” option:
rule before 2 request INVITE sip-header To Modify “(.*)” “\1;temp=abc”
before
voice class sip-profiles 1
option
rule 1 request INVITE sip-header Contact Modify “(.*)” “\1;temp=xyz”
rule 2 request INVITE sip-header To Modify “(.*)” “\1;temp=abc”
The new rule has
rule 3 request INVITE sip-header Supported Add “Supported: ”
been inserted
between #1 and
#3
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
49
Multiple Incoming Patterns Under
Same Incoming/Outgoing Dial-peer
Site A
Site B
Site C
voice class e164-pattern-map 300
e164 919200200.
e164 510100100.
(919)200-2000
e164 408100100.
dial-peer voice 1 voip
description Inbound DP via Calling
incoming calling e164-pattern-map 300
(408)100-1000
codec g729r8
(510)100-1000
G729 Sites
A
Module 8
Provides the ability to combine
multiple incoming called OR calling
numbers on a single inbound voip
dial-peer, reducing the total number
of inbound voip dial-peers required
with the same routing capability
Up to 5000 entries in a text file
SP SIP Trunk
SIP Trunk
IP PSTN
CUBE
Site A
(919)200-2010
Site B
(510)100-1010
voice class e164-pattern-map 400
url flash:e164-pattern-map.cfg
dial-peer voice 2 voip
description Outbound DP via Called
(408)100-1010
Site C
destination e164-pattern-map 400
codec g711ulaw
G711 Sites
LTRCOL-2310
! This is an example of the contents
of E164 patterns text file stored
in flash:e164-pattern-map.cfg
9192002010
5101001010
4081001010
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50
Module 9
Destination Server Group
Supports multiple destinations (session targets) be defined in a group and
applied to a single outbound dial-peer
• Once an outbound dial-peer is selected to route an outgoing call, multiple
destinations within a server group will be sorted in either round robin or
preference [default] order
• This reduces the need to configure multiple dial-peers with the same
capabilities but different destinations. E.g. Multiple subscribers in a cluster
•
voice class server-group 1
hunt-scheme {preference | round-robin}
ipv4 1.1.1.1 preference 5
ipv4 2.2.2.2
ipv4 3.3.3.3 port 3333 preference 3
ipv6 2010:AB8:0:2::1 port 2323 preference 3
ipv6 2010:AB8:0:2::2 port 2222
dial-peer voice 100 voip
description Outbound DP
destination-pattern 1234
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
session server-group 1
* DNS target not supported in server group
LTRCOL-2310
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51
Module 10
External/PSTN Call Recording Options
•
•
CUBE Controlled (Dial-peer based SIPREC)
•
Based on SIPREC (RFC 6341, 7245, Metadata-draft-17, Protocol-draft-15), CUBE
sends metadata in XML format
•
Dial-peer controlled, IP-PBX independent
•
Source of recorded media (RTP only) is always CUBE (External calls only). For
SRTP-RTP calls, apply media forking CLI on the RTP leg only.
•
Records both audio and video calls and supported with CUBE HA (Inbox or box-2box)
CUCM NBR (Network Based Recording)
•
CUCM Controlled, requires CUCM 10+ and UC Services API be enabled on CUBE
•
Recording triggered by CUCM and this mode records only Audio calls
•
Source of Recorded Media can be CUBE or Endpoint (BiB), CUBE as source
desired for PSTN calls
LTRCOL-2310
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52
CUBE Controlled Recording Option – Media Forking
Dial-peer based – Open Recording Architecture (ORA)
• CUBE sets up a stateful SIP session with MediaSense server
• After SIP dialog established, CUBE forks the RTP and sends it for MediaSense to record
• With XE 3.10.1, Video calls supported and CUBE HA for audio calls
198.18.133.186
Cisco MediaSense
(authentication disabled w/o UCM)
SIP
Cisco Proprietary Metadata
A
SIP
SIP
SP SIP
RTP
• Call agent
independent
• Configured on a per
Dial-peer level to fork
RTP
CUBE
media class 10
recorder parameter
media-recording 1050
dial-peer voice 101 voip
RTP
dial-peer voice 1050 voip
description dial-peer pointing to MediaSense
Needs to
match
description dial-peer that needs to be forked
session protocol sipv2
media-class 10
destination-pattern 9999 ! Dummy
session protocol sipv2
session transport tcp
session target ipv4:198.18.133.186
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
CUBE Controlled Recording Option - SIPREC
Dial-peer based – SIPREC Standard
Module 10
• SIP is used as a protocol between CUBE and the recording server, where CUBE acts as the recording
client and any third party recorder acts as the recording server
• Along with SDP, metadata information is passed by CUBE to the recording server in XML format
• Metadata includes the
communication session details of
audio or video calls and also
identifies the participants of the call
Recorder
SIPREC Compliant Recorder
XML Metadata
A
SIP
SIP
SP SIP
RTP
• SIP Profiles can
additionally be used to
forward 3rd party IP PBX
Call Identifier to the
Recorder for Correlation
CUBE
media class 10
recorder parameter siprec
media-recording 1050
dial-peer voice 101 voip
RTP
dial-peer voice 1050 voip
description dial-peer pointing to MediaSense
Needs to
match
description dial-peer that needs to be forked
session protocol sipv2
media-class 10
destination-pattern 9999 ! Dummy
session protocol sipv2
session transport tcp
session target ipv4:<Recorder_IP>
! Bind on this DP mandatory
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
CUCM (10.X or later) Controlled Recording
UC Services API – Network Based Recording
3.
Module 10
1. Enable HTTP on IOS
Gateway/CUBE Recording
Enabled
ip http server
http client persistent
2. Enable the API on IOS
4.
1.
2.
uc wsapi
source-address [IP_Address_of_CUBE]
3. Enable XMF service within the API
5.
provider xmf
remote-url 1 http://CUCM:8090/ucm_xmf
no shutdown
[1] – [3]: An external call is answered by user with IP phone
[4] – [5]: CUCM sends forking request over HTTP to CUBE, which
sends two media streams towards the Recording Server
• Recording not preserved on failover in CUBE HA
• Selective Recording, Mobile/SNR/MVA Calls
• Recording Call Preservation
Now Supports Inbound CVP (Survivability.tcl) Call Recording
[IOS 15.6(1)T, IOS-XE 3.17]
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved.
Cisco Public
55
Module 11
Secure SIP Trunks with CUBE
LAN
WAN
Gig0/0
Gig0/1
TCP/UDP
RTP
SIP TLS
CUBE
SP IP
Network
198.18.133.3
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
56
Module 12
Branch CUBE Deployment with SRST Provisioned
Branch with Unified SRST Provisioned
on the same platform as CUBE
Unified CM
LAN Dial-Peers
WAN Dial-Peers
CUBE
Gig0/0
Data
Center
PSTN
Enterprise
IP WAN
Gig0/1
RTP
SIP - Trunkside
SIP - Lineside
SIP Endpoints
Enterprise
LAN
ITSP
LTRCOL-2310
WAN (SIP Provider)
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
57
Branch losing connectivity to Unified CM across
the Enterprise WAN
Branch with no WAN connectivity
Unified CM
LAN Dial-Peers
WAN Dial-Peers
CUBE
Gig0/0
Data
Center
Enterprise
IP WAN
PSTN
Gig0/1
RTP
SIP - Trunkside
SIP - Lineside
Enterprise
LAN
ITSP
LTRCOL-2310
WAN (SIP Provider)
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
58
Branch Operating in SRST mode with CUBE SIP
Trunk for PSTN calling
Branch in SRST mode
Unified CM
LAN Dial-Peers (Busied
Out)
WAN Dial-Peers (Unchanged)
SRST
Gig0/0
Data
Center
Enterprise
IP WAN
PSTN
Gig0/1
Voice Register
Pool / DNs
RTP
SIP - Trunkside
SIP - Lineside
Enterprise
LAN
ITSP
LTRCOL-2310
WAN (SIP Provider)
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
59
Overview: Lab Overview
Lab – Network Setup
LAN
CUCM
WAN
CUBE
Gig0/0 – 198.18.128.11
Gig0/1 – 10.1.40.POD
SP IP
Network
CUBE
Cisco MediaSense
198.18.133.3
ITSP SIP Trunk
10.1.40.11
198.18.133.186
Windows Work Station
1 or 3 with Jabber
SoftPhone
Internet
IP – 198.18.133.38/39
Phone# +1(408)944-29DN
LTRCOL-2310
Classroom PC
OR
Your PC
with AnyConnect
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61
Lab – Access
•
Use Cisco AnyConnect VPN to connect to Cisco dCloud infrastructure. VPN
connection details are provided in the printed information sheet for your pod.
•
Once VPN’d in, connect to the following workstation using the local RDP client on
your laptop.
•
Workstation 3: 198.18.133.38, Username: dcloud\mcheng, Password: C1sco12345
•
Once connected to WKST3, launch the Cisco Jabber for Windows client by doubleclicking the desktop icon
•
Login to the Jabber client by entering the password: C1sco12345 and clicking Sign
In. Note that the username field is already pre-populated with our WKST3 user’s
login username: mcheng.
•
Access CUBE using the PuTTY Client and CUCM via a browser. IP
Addresses and credentials are in your respective pod sheets
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
62
Key Takeaways
•
It is a manageable transition from existing TDM based networks to SIP networks using these
network design techniques
•
Enterprise SBC (Cisco Unified Border Element - CUBE) is an essential component of a UC
solution providing;
•
Security, Session Management, Interworking, Demarcation
•
Over 30,000 Enterprise customers all over the Globe
•
Proven interoperability with 3rd party PBX vendors and different service providers around the world (more than
165 countries)
•
Now is the time to deploy SIP Trunking in either a Centralized or a Distributed solution to save
money, simplify your topology and setup your infrastructure for future services
•
Complete feature Presentations, Lab Guide, Free Hands-on Lab access & Application Notes :
https://cisco.box.com/cube
LTRCOL-2310
© 2019 Cisco and/or its affiliates. All rights reserved. Cisco Public
63
Cisco Webex Teams
Questions?
Use Cisco Webex Teams (formerly Cisco Spark)
to chat with the speaker after the session
How
1 Find this session in the Cisco Events Mobile App
2 Click “Join the Discussion”
3 Install Webex Teams or go directly to the team space
4 Enter messages/questions in the team space
cs.co/ciscolivebot#LTRCOL-2310
LTRCOL-2310
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64
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session survey
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Don’t forget: Cisco Live sessions will be available for viewing
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LTRCOL-2310
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Showcase
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self-paced
labs
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engineer
1:1
meetings
LTRCOL-2310
Related
sessions
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Thank you