SIP-to-SIP Extended Feature Functionality for Session Border Controllers
The SIP-to-SIP Extended Feature Functionality for Session Border Controllers (SBCs) enables the SIP-to-SIP functionality to conform with RFC 3261 to interoperate with SIP User Agents (UAs). The SIP-to-SIP Extended Feature Functionality includes:
Call Admission Control (based on CPU, memory, and total calls)
Delayed Media Call
ENUM support
Configuring SIP Error Message Pass Through
Interoperability with Cisco Unified Communications Manager 5.0 and BroadSoft
Your software release may not support all the features documented in this module. For the latest caveats and feature information, see
Bug Search Tool and the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the feature information table at the end of this module.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to
www.cisco.com/go/cfn. An account on Cisco.com is not required.
Prerequisites for SIP-to-SIP Extended Feature Functionality for Session Border Controllers
Cisco Unified Border Element
Cisco IOS Release 12.4(6)T or a later release must be installed and running on your Cisco Unified Border Element.
Cisco Unified Border Element (Enterprise)
Cisco IOS XE Release 3.1S or a later release must be installed and running on your Cisco ASR 1000 Series Router.
Modem Passthrough over VoIP
The Modem Passthrough over VoIP feature provides the transport of modem signals through a packet network by using pulse code modulation (PCM) encoded packets.
Prerequisites for the Modem Passthrough over VoIP Feature
VoIP enabled network.
Cisco IOS Release 12.1(3)T must run on the gateways for the Modem Passthrough over VoIP feature to work.
Network suitability to pass modem traffic. The key attributes are packet loss, delay, and jitter. These characteristics of the network can be determined by using the Cisco IOS feature Service Assurance Agent.
Cisco Unified Border Element
Cisco IOS Release 12.4(6)T or a later release must be installed and running on your Cisco Unified Border Element.
Cisco Unified Border Element (Enterprise)
Cisco IOS XE Release 3.3S or a later release must be installed and running on your Cisco ASR 1000 Series Router.
Restrictions for the Modem Passthrough over VoIP Feature
Cisco Unified Border Element (Enterprise)
If call started as g729, upon modem tone (2100Hz) detection both the outgoing gateway (OGW) and the trunking gateway (TGW) will genearate NSE packets towards peer side and up speed to g711 as Cisco UBE(Enterprise) passes these packets to the peer side.
Note
That OGW and TGW display the new codec, but the Cisco UBE (Enterprise) continues to show the original codec g729 in the show commands.
Information about Configuring Modem Passthrough over VoIP
The Modem Passthrough over VoIP feature performs the following functions:
Represses processing functions like compression, echo cancellation, high-pass filter, and voice activity detection (VAD).
Issues redundant packets to protect against random packet drops.
Provides static jitter buffers of 200 milliseconds to protect against clock skew.
Discriminates modem signals from voice and fax signals, indicating the detection of the modem signal across the connection, and placing the connection in a state that transports the signal across the network with the least amount of distortion.
Reliably maintains a modem connection across the packet network for a long duration under
normal network conditions.
For further details, the functions of the Modem Passthrough over VoIP feature are described in the following sections.
Modem Tone Detection
The gateway is able to detect modems at speeds up to V.90.
Passthrough Switchover
When the gateway detects a data modem, both the originating gateway and the terminating gateway roll over to G.711. The roll over to G.711 disables the high-pass filter, disables echo cancellation, and disables VAD. At the end of the modem call, the voice ports revert to the prior configuration and the digital signal processor (DSP) goes back to the state before switchover. You can configure the codec by selecting the
g711alaw or
g711ulaw option of the
codec command.
You can enable payload redundancy so that the Modem Passthrough over VoIP switchover causes the gateway to emit redundant packets.
Packet Size
When redundancy is enabled, 10-ms sample-sized packets are sent. When redundancy is disabled, 20-ms sample-sized packets are sent.
Clock Slip Buffer Management
When the gateway detects a data modem, both the originating gateway and the terminating gateway switch from dynamic jitter buffers to static jitter buffers of 200-ms depth. The switch from dynamic to static is to compensate for Public Switched Telephone Network (PSTN) clocking differences at the originating gateway and the terminating gateway. At the conclusion of the modem call, the voice ports revert to dynamic jitter buffers.
The figure below illustrates the connection from the client modem to a MICA technologies modem network access server (NAS).
Figure 1. Modem Passthrough Connection
How to Configure Modem Passthrough over VoIP
You can configure the Modem Passthrough over VoIP feature on a specific dial peer in two ways, as follows:
Globally in the voice-service configuration mode
Individually in the dial-peer configuration mode on a specific dial peer
By default, modem passthrough over VoIP capability and redundancy are disabled.
Tip
You need to configure modem passthrough in both the originating gateway and the terminating gateway for the Modem Passthrough over VoIP feature to operate. If you configure only one of the gateways in a pair, the modem call will not connect successfully.
Redundancy can be enabled in one or both of the gateways. When only a single gateway is configured for redundancy, the other gateway receives the packets correctly, but does not produce redundant packets.
See the following sections for the Modem Passthrough over VoIP feature. The two configuration tasks can configure separately or together. If both are configured, the dial-peer configuration takes precedence over the global configuration. Consequently, a call matching a particular dial-peer will first try to apply the modem passthrough configuration on the dial-peer. Then, if a specific dial-peer is not configured, the router will use the global configuration:
For the Modem Passthrough over VoIP feature to operate, you need to configure modem passthrough in both the originating gateway and the terminating gateway so that the modem call matches a voip dial-peer on the gateway.
The default behavior for the voice-service configuration mode is
nomodempassthrough. This default behavior implies that modem passthrough is disabled for all dial peers on the gateway by default.
When using the
voiceservicevoip and
modempassthroughnse commands on a terminating gateway to globally set up fax or modem passthrough with NSEs, you must also ensure that each incoming call will be associated with a VoIP dial peer to retrieve the global fax or modem configuration. You associate calls with dial peers by using the
incomingcalled-number command to specify a sequence of digits that incoming calls can match.
To configure the Modem Passthrough over VoIP feature for all the connections of a gateway, use the following commands beginning in global configuration mode:
Configures the Modem Passthrough over VoIP feature The default behavior is
nomodempassthrough.
The payload type is an optional parameter for the
nse keyword. Use the same
payload-typenumber for both the originating gateway and the terminating gateway. The
payload-typenumbercan be set from 96 to 119. If you do not specify the
payload-typenumber, the
numberdefaults to 100. When the
payload-type is 100, and you use the
showrunning-configcommand, the
payload-type parameter does not appear.
Use the same codec type for both the originating gateway and the terminating gateway.
g711ulaw codec is required for T1, and
g711alaw codec is required for E1.
The
redundancy keyword is an optional parameter for sending redundant packets for modem traffic.
The
maximum-sessions keyword is an optional parameter for the
redundancykeyword. This parameter determines the maximum simultaneous modem passthrough sessions with
redundancy.
Step 4
exit
Example:
Device(conf-voi-serv)# exit
Exits voice-service configuration mode.
Step 5
exit
Example:
Device(config)# exit
Exits global configuration mode.
Configuring Modem Passthrough over VoIP for a Specific Dial Peer
To enable Modem Passthrough on the VoIP dial peers on both the originating and terminating gateway, configure modem passthrough globally or explicitly on the dial peer.
For modem passthrough to operate, you must define VoIP dial peers on both gateways to match the call, for example, by using a destination pattern or an incoming called number. The modem passthrough parameters associated with those dial peers then will apply to the call.
Note
When modem passthrough is configured individually for a specific dial peer, that configuration for the specific dial peer takes precedence over the global configuration.
To configure the Modem Passthrough over VoIP feature for a specific dial peer, use the following commands beginning in global configuration mode:
Configures the Modem Passthrough over VoIP feature for a specific dial peer. The default behavior for the Modem Passthrough for VoIP feature in dial-peer configuration mode is
modempassthroughsystem. As required, the gateway defaults to
nomodempassthrough.
When the
system keyword is enabled, the following parameters are not available:
nse,
payload-type,
codec, and
redundancy. Instead the values from the global configuration are used.
The payload type is an optional parameter for the
nse keyword. Use the same
payload-typenumber for both the originating gateway and the terminating gateway. The
payload-typenumbercan be set from 96 to 119. If you do not specify the
payload-typenumber, the
numberdefaults to 100. When the
payload-type is 100, and you use the
showrunning-configcommand, the
payload-type parameter does not appear.
Use the same codec type for both the originating gateway and the terminating gateway.
g711ulaw codec is required for T1, and
g711alaw codec is required for E1.
The
redundancy keyword is an optional parameter for sending redundant packets for modem traffic.
Step 4
exit
Example:
Device(config-dial-peer)# exit
Exits dial-peer configuration mode and returns to the global configuration mode.
Step 5
exit
Example:
Device(config)# exit
Exits global configuration mode.
Troubleshooting Tips
To troubleshoot the Modem Passthrough over VoIP feature, perform the following steps:
Make sure that you can make a voice call.
Make sure that Modem Passthrough over VoIP is configured on both the originating gateway and the terminating gateway.
Make sure that both the originating gateway and the terminating gateway have the same named signaling event (NSE) payload-typenumber.
Make sure that both the originating gateway and the terminating gateway have the same maximum-sessionsvalue when the two gateways are configured in the voice-service configuration mode.
Use the debugvtspdsp and debugvtspsession commands to debug a problem.
Verifying Modem Passthrough over VoIP
To verify that the Modem Passthrough over VoIP feature is enabled, perform the following steps:
SUMMARY STEPS
1. Enter the
showrun command to verify the configuration.
2. Enter the
showdial-peervoice command to verify that Modem Passthrough over VoIP is enabled.
DETAILED STEPS
Step 1
Enter the
showrun command to verify the configuration.
Step 2
Enter the
showdial-peervoice command to verify that Modem Passthrough over VoIP is enabled.
Monitoring and Maintaining Modem Passthrough over VoIP
To monitor and maintain the Modem Passthrough over VoIP feature, use the following commands in privileged EXEC mode:
Command
Purpose
Device# show call active voice brief
Displays information for the active call table or displays the voice call history table. The brief option displays a truncated version of either option.
Device# show dial-peer voice 15 summary
Displays configuration information for dial peers. The
number argument specifies a specific dial peer from 1 to 32767. The summary option displays a summary of all dial peers.
Configuration Examples
The following is sample configuration for the Modem Passthrough over VoIP feature:
version 12.1
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
voice service voip
modem passthrough nse codec g711ulaw redundancy maximum-session 5
!
!
resource-pool disable
!
!
!
!
!
ip subnet-zero
ip ftp source-interface Ethernet0
ip ftp username lab
ip ftp password lab
no ip domain-lookup
!
isdn switch-type primary-5ess
cns event-service server
!
!
!
!
!
mta receive maximum-recipients 0
!
!
controller T1 0
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24
!
controller T1 1
shutdown
clock source line secondary 1
!
controller T1 2
shutdown
!
controller T1 3
shutdown
!
!
!
interface Ethernet0
ip address 1.1.2.2 255.0.0.0
no ip route-cache
no ip mroute-cache
!
interface Serial0:23
no ip address
encapsulation ppp
ip mroute-cache
no logging event link-status
isdn switch-type primary-5ess
isdn incoming-voice modem
no peer default ip address
no fair-queue
no cdp enable
no ppp lcp fast-start
!
interface FastEthernet0
ip address 26.0.0.1 255.0.0.0
no ip route-cache
no ip mroute-cache
load-interval 30
duplex full
speed auto
no cdp enable
!
ip classless
ip route 17.18.0.0 255.255.0.0 1.1.1.1
no ip http server
!
!
!
!
voice-port 0:D
!
dial-peer voice 1 pots
incoming called-number 55511..
destination-pattern 020..
direct-inward-dial
port 0:D
prefix 020
!
dial-peer voice 2 voip
incoming called-number 020..
destination-pattern 55511..
modem passthrough nse codec g711ulaw redundancy
session target ipv4:26.0.0.2
!
!
line con 0
exec-timeout 0 0
transport input none
line aux 0
line vty 0 4
login
!
!
end
Feature Information for SIP-to-SIP Extended Feature Functionality for Session Border Controllers
The following table provides release information about the feature or features described in this module. This table lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to
www.cisco.com/go/cfn. An account on Cisco.com is not required.
Table 1 Feature Information for Configuring SIP-to-SIP Extended Feature Functionality for Session Border Controllers
Feature Name
Releases
Feature Information
SIP-to-SIP Extended Feature Functionality for Session Border Controllers
12.4(6)T
The SIP-to-SIP Extended Feature Functionality for Session Border Controllers (SBCs) enables the SIP-to-SIP functionality to conform with RFC 3261 to interoperate with SIP User Agents (UAs).
In Cisco IOS Release 12.4(6)S, this feature was implemented on the Cisco Unified Border Element
The following commands were introduced or modified:
modempassthrough(dial-peer);
modempassthrough(voice-service);
showcallactivevoicevoice;
showcallhistoryvoicevoice;
showdial-peervoice;
voiceservice.
SIP-to-SIP Extended Feature Functionality for Session Border Controllers
Cisco IOS XE Release 3.1S
Cisco IOS XE Release 3.3S
The SIP-to-SIP Extended Feature Functionality for Session Border Controllers (SBCs) enables the SIP-to-SIP functionality to conform with RFC 3261 to interoperate with SIP User Agents (UAs).
In Cisco IOS Release 12.4(6)S, this feature was implemented on the Cisco Unified Border Element (Enterprise).
The following commands were introduced or modified:
modempassthrough(dial-peer);
modempassthrough(voice-service);
showcallactivevoicevoice;
showcallhistoryvoicevoice;
showdial-peervoice;
voiceservice.