The documentation set for this product strives to use bias-free language. For the purposes of this documentation set, bias-free is defined as language that does not imply discrimination based on age, disability, gender, racial identity, ethnic identity, sexual orientation, socioeconomic status, and intersectionality. Exceptions may be present in the documentation due to language that is hardcoded in the user interfaces of the product software, language used based on RFP documentation, or language that is used by a referenced third-party product. Learn more about how Cisco is using Inclusive Language.
This document describes VoIP in IPv6 (VoIPv6), a feature that adds IPv6 capability to existing VoIP features. This feature adds dual-stack (IPv4 and IPv6) support on voice gateways and media termination points (MTPs), IPv6 support for Session Initiation Protocol (SIP) trunks, and support for Skinny Client Control Protocol (SCCP)-controlled analog voice gateways. In addition, the Session Border Controller (SBC) functionality of connecting a SIP IPv4 or H.323 IPv4 network to a SIP IPv6 network is implemented on a Cisco UBE to facilitate migration from VoIPv4 to VoIPv6.
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.
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.
Cisco Express Forwarding for IPv6 must be enabled.
Virtual routing and forwarding (VRF) is not supported in IPv6 calls.
The following are the restrictions for Cisco UBE features:
Supports only Early Offer (EO)–Early Offer (EO) and Delayed Offer (DO)–Delayed Offer (DO) call flows.
Delayed Offer–Early Offer call flow falls back to Delayed Offer–Delayed Offer call flow.
Supplementary services are not supported on SDP Pass-Through.
Transcoding and DTMF interworking are not supported.
![]() Note | The above SDP Pass–Through restrictions are applicable for both IPv4 and IPv6. |
SDP Pass–Through does not support the dual-stack functionality.
ANAT call flows does not support IPv4-to-IPv6 and IPv6-to-IPv4 Media interworking.
Media Anti–Trombone is not enabled if the initial call before tromboning is in Flow–Around (FA) mode.
Media Anti–Trombone supports only symmetric media address type interworking (IPv4-IPv4 or IPv6-IPv6 media) with or without ANAT.
Does not provide support for IPv4-IPv6 interworking cases with or without ANAT because Cisco UBE cannot operate in FA mode post tromboning.
The Session Initiation Protocol (SIP) is an alternative protocol developed by the Internet Engineering Task Force (IETF) for multimedia conferencing over IP.
A SIP User Agent (UA) operates in one of the following three modes:
IPv4-only: Communication with only IPv6 UA is unavailable.
IPv6-only: Communication with only IPv4 UA is unavailable.
Dual-stack: Communication with only IPv4, only IPv6 and dual-stack UAs are available.
Dual-stack SIP UAs use Alternative Network Address Transport (ANAT) grouping semantics:
Includes both IPv4 and IPv6 addresses in the Session Description Protocol (SDP).
Is automatically enabled in dual-stack mode (can be disabled if required).
Requires media to be bound to an interface that have both IPv4 and IPv6 addresses.
Described in RFC 4091 and RFC 4092 (RFC 5888 describes general SDP grouping framework).
SIP UAs use “sdp-anat” option tag in the Required and Supported SIP header fields:
Early Offer (EO) INVITE using ANAT semantics places “sdp-anat” in the Require header.
Delayed Offer (DO) INVITE places “sdp-anat” in the Supported header.
SDP may or may not use ANAT semantics:
When ANAT is used, media addresses in SDP are chosen from the interface media that is configured. When ANAT is not used, media addresses in SDP are chosen from the interface media that is configured OR based on the best address to reach the destination signaling address (when no media bind is configured).
Cisco UBE in VoIPv6 adds IPv6 capability to VoIP features. This feature adds dual-stack support on voice gateways, IPv6 support for SIP trunks, support for SCCP-controlled analog voice gateways, support for real-time control protocol (RTCP) pass-through, and support for T.38 fax over IPv6.
“Configuring Cisco IOS Gateways” section in the Deploying IPv6 in Unified Communications Networks with Cisco Unified Communications Manager
“Trunks” section in Deploying IPv6 in Unified Communications Networks with Cisco Unified Communications Manager
“SCCP-controlled analog voice gateways” section in the SCCP Controlled Analog (FXS) Ports with Supplementary Features in Cisco IOS Gateways
“RTCP Pass-Through” section in Cisco UBE RTCP Voice Pass-Through for IPv6
“T.38 fax over IPv6” section in Fax, Modem, and Text Support over IP Configuration Guide
Support has been added for audio calls in media Flow–Through (FT) and Flow–Around (FA) modes, High Density (HD) transcoding, Local Transcoding Interface (LTI), along with Voice Class Codec (VCC) support, support for Hold/Resume, REFER, re-INVITE, 302 based services, and support for media anti-trombone have been added to Cisco UBE.
Cisco UBE being a signaling proxy processes all signaling messages for setting up media channels. This enables Cisco UBE to affect the flow of media packets using the media flow-through and the media flow-around modes.
Media Flow-Through (FT): In a media flow–through mode, between two endpoints, both signaling and media flows through the IP-to-IP Gateway (IPIP GW). The IPIP GW performs both signaling and media interworking between H.323/SIP IPv4 and SIP IPv6 networks.
Media Flow-Around (FA): Media flow–around provides the ability to have a SIP video call whereby signaling passes through Cisco UBE and media pass directly between endpoints bypassing the Cisco UBE.
Assisted RTCP (RTCP Keepalive): Assisted Real-time Transport Control Protocol (RTCP) enables Cisco UBE to generate RTCP keepalive reports on behalf of endpoints; however, endpoints, such as second generation Cisco IP phones (7940/7960) and Nortel Media Gateways (MG 1000T) do not generate any RTCP keepalive reports. Assisted RTCPs enable customers to use Cisco UBE to interoperate between endpoints and call control agents, such as Microsoft OCS/Lync so that RTCP reports are generated to indicate session liveliness during periods of prolonged silence, such as call hold or call on mute.
The assisted RTCP feature helps Cisco UBE to generate standard RTCP keepalive reports on behalf of endpoints. RTCP reports determine the liveliness of a media session during prolonged periods of silence, such as a call on hold or a call on mute.
SDP Pass–Through: SDP is configured to pass through transparently at the Cisco UBE, so that both the remote ends can negotiate media independently of the Cisco UBE.
Flow-through—Cisco UBE plays no role in the media negotiation, it blindly terminates and re-originates the RTP packets irrespective of the content type negotiated by both the ends. This supports address hiding and NAT traversal.
Flow-around—Cisco UBE neither plays a part in media negotiation, nor does it terminate and re-originate media. Media negotiation and media exchange is completely end-to-end.
For more information, refer to the “Configurable Pass-through of SIP INVITE Parameters” section in the Cisco Unified Border Element SIP Support Configuration Guide.
UDP Checksum for IPv6: User Datagram Protocol (UDP) checksums provide data integrity for addressing different functions at the source and destination of the datagram, when a UDP packet originates from an IPv6 node.
IP Toll Fraud:The IP Toll Fraud feature checks the source IP address of the call setup before routing the call. If the source IP address does not match an explicit entry in the configuration as a trusted VoIP source, the call is rejected.
For more information, refer to the “Configuring Toll Fraud Prevention” section in the Cisco Unified Communications Manager Express System Administrator Guide.
RTP Port Range: Provides the capability where the port range is managed per IP address range. This features solves the problem of limited number of rtp ports for more than 4000 calls. It enables combination of an IP address and a port as a unique identification for each call.
Hold/Resume: Cisco UBE supports supplementary services such as Call Hold and Resume. An active call can be put in held state and later the call can be resumed.
For more information, refer to the “Configuring Call Hold/Resume for Shared Lines for Analog Ports” section in Supplementary Services Features for FXS Ports on Cisco IOS Voice Gateways Configuration Guide.
Call Transfer (re-INVITE, REFER): Call transfer is used for conference calling, where calls can transition smoothly between multiple point-to-point links and IP level multicasting.
For more information, refer to the “Configurable Pass-through of SIP INVITE Parameters” section in the Cisco Unified Border Element SIP Support Configuration Guide.
For more information, refer to the “ Configuring Call Transfer and Forwarding” section in Cisco Unified Communications Manager Express System Administrator Guide.
Media Antitrombone: Antitromboning is a media signaling service in SIP entity to overcome the media loops. Media Trombones are media loops in a SIP entity due to call transfer or call forward. Media loops in Cisco UBE are not detected because Cisco UBE looks at both call types as individual calls and not calls related to each other.
Antitrombone service has to be enabled only when no media interworking is required in both legs. Media antitrombone is supported only when the initial call is in IPv4 to IPv4 or IPv6 to IPv6 mode only.
For more information, refer to the “Configuring Media Antitrombone” section in the Cisco Unified Border Element Protocol-Independent Features and Setup Configuration Guide.
RE-INVITE Consumption: The Re-INVITE/UPDATE consumption feature helps to avoid interoperability issues by consuming the mid-call Re-INVITEs/UPDATEs from Cisco UBE. As Cisco UBE blocks RE-INVITE / mid-call UPDATE, remote participant is not made aware of the SDP changes, such as Call Hold, Call Resume, and Call transfer.
For more information, refer to the “Cisco UBE Mid-call Re-INVITE/UPDATE Consumption” section in the Cisco Unified Border Element Protocol-Independent Features and Setup Configuration Guide.
Address Hiding: The address hiding feature ensures that the Cisco UBE is the only point of signaling and media entry/exit in all scenarios. When you configure address-hiding, signaling and media peer addresses are also hidden from the endpoints, especially for supplementary services when the Cisco UBE passes REFER/3xx messages from one leg to the other.
For more information, refer to the “Configuring Address Hiding” section in the SIP-to-SIP Connections on a Cisco Unified Border Element.
Header Passing: Header Pass through enables header passing for SIP INVITE, SUBSCRIBE and NOTIFY messages; disabling header passing affects only incoming INVITE messages. Enabling header passing results in a slight increase in memory and CPU utilization.
For more information, refer to the “SIP-to-SIP Connections on a Cisco Unified Border Element” section in the SIP-to-SIPConnections on Cisco Unified Border Element.
Refer–To Passing: The Refer-to Passing feature is enabled when you configure refer-to-passing in Refer Pass through mode and the supplementary service SIP Refer is already configured. This enables the received refer-to header in Refer Pass through mode to move to the outbound leg without any modification. However, when refer-to-passing is configured in Refer Consumption mode without configuring the supplementary-service SIP Refer, the received Refer-to URI is used in the request-URI of the triggered invite.
For more information, refer to the “Configuring Support for Dynamic REFER Handling on Cisco UBE” section in the Cisco Unified Border Element SIP Configuration Guide.
Error Pass-through: The SIP error message pass through feature allows a received error response from one SIP leg to pass transparently over to another SIP leg. This functionality will pass SIP error responses that are not yet supported on the Cisco UBE or will preserve the Q.850 cause code across two sip call-legs.
For more information, refer to the “Configuring SIP Error Message Passthrough” section in the Cisco Unified Border Element SIP Support Configuration Guide.
For more information, refer to the “SIP UPDATE Message per RFC 3311” section in the Cisco Unified Border Element SIP Support Configuration Guide.
SIP OPTIONS Ping: The OPTIONS ping mechanism monitors the status of a remote Session Initiation Protocol (SIP) server, proxy or endpoints. Cisco UBE monitors these endpoints periodically.
For more information, refer to the “Cisco UBE Out-of-dialog OPTIONS Ping for Specified SIP Servers or Endpoints” section in the Configuration of SIP Trunking for PSTN Access (SIP-to-SIP) Configuration Guide.
Configurable Error Response Code in OPTIONS Ping: Cisco UBE provides an option to configure the error response code when a dial peer is busied out because of an Out-of-Dialog OPTIONS ping failure.
For more information, refer to the “Configuring an Error Response Code upon an Out-of-Dialog OPTIONS Ping Failure” section in the Cisco Unified Border Element SIP Support Configuration Guide.
SIP Profiles: SIP profiles create a set of provisioning properties that you can apply to SIP trunk.
Dynamic Payload Type Interworking (DTMF and Codec Packets): The Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP-to-SIP Calls feature provides dynamic payload type interworking for dual tone multifrequency (DTMF) and codec packets for Session Initiation Protocol (SIP) to SIP calls. The Cisco UBE interworks between different dynamic payload type values across the call legs for the same codec. Also, Cisco UBE supports any payload type value for audio, video, named signaling events (NSEs), and named telephone events (NTEs) in the dynamic payload type range 96 to 127.
For more information, refer to the “Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP-to-SIP Calls” section in the Cisco Unified Border Element (Enterprise) Protocol-Independent Features and Setup Configuration Guide.
Audio Transcoding using Local Transcoding Interface (LTI): Local Transcoding Interface (LTI) is an interface created to remove the requirement of SCCP client for Cisco UBE transcoding.
For information, refer to Cisco Unified Border Element 9.0 Local Transcoding Interface (LTI).
Voice Class Codec (VCC) with or without Transcoding: The Voice Class Codec feature supports basic and all Re-Invite based supplementary services like call-hold/resume, call forward, call transfer, where if any mid-call codec changes, Cisco UBE inserts/removes/modifies the transcoder as needed.
Support for negotiation of an Audio Codec on each leg of a SIP–SIP call on the Cisco UBE feature supports negotiation of an audio codec using the Voice Class Codec (VCC) infrastructure on Cisco UBE.
VCC supports SIP-SIP calls on Cisco UBE and allows mid-call codec change for supplementary services.
Session Initiation Protocol (SIP) is a simple, ASCII-based protocol that uses requests and responses to establish communication among the various components in the network and to ultimately establish a conference between two or more endpoints.
In addition to the already existing features that are supported on IPv4 and IPv6, the SIP Voice Gateways support the following features:
History–Info: The SIP History–info Header Support feature provides support for the history-info header in SIP INVITE messages only. The SIP gateway generates history information in the INVITE message for all forwarded and transferred calls. The history-info header records the call or dialog history. The receiving application uses the history-info header information to determine how and why the call has reached it.
For more information, refer to the “SIP History INFO” section in the Cisco Unified Border Element (Enterprise) SIP Support Configuration Guide.
Handling 181/183 Responses with/without SDP: The Handling 181/183 Responses with/without SDP feature provides support for SIP 181 (Call is Being Forwarded) and SIP 183 (Session Progress) messages either globally or on a specific dial-peer. Also, you can control when the specified SIP message is dropped based on either the absence or presence of SDP information.
For more information, refer to “SIP–Enhanced 180 Provisional Response Handling” section in the Cisco Unified Border Element Configuration Guide.
Limiting the Rate of Incoming SIP Calls per Dial-Peer (Call Spike): The call rate-limiting feature for incoming SIP calls starts working after a switch over in a SIP call. The rate–limiting is done for new calls received on the new Active. The IOS timers that track the call rate limits runs on active and standby mode and does not require any checkpoint. However, some statistics for calls rejected requires to be checked for the show commands to be consistent before and after the switchover.
PPI/PAI/Privacy and RPID Passing: For incoming SIP requests or response messages, when the PAI or PPI privacy header is set, the SIP gateway builds the PAI or PPI header into the common SIP stack, thereby providing support to handle the call data present in the PAI or PPI header. For outgoing SIP requests or response messages, when the PAI or PPI privacy header is set, privacy information is sent using the PAI or PPI header.
For more information, refer to the “Support for PAID PPID Privacy PCPID and PAURI Headers on Cisco UBE” section in the Cisco Unified Border Element SIP Support Configuration Guide.
SIP VMWI for FXS phones: SIP provides visible message waiting indication (VMWI) on FXS phones. This feature provides users with the option to enable one message waiting indication (MWI): audible, visible, or both. The VMWI mechanism uses SIP Subscribe or Notify to get MWI updates from a virtual machine (VM) system, and then forwards updates to the FXS phone on the port.
For more information, refer to the “Configuring SIP MWI Features” section in the SIP Configuration Guide.
SIP Session timer (RFC 4028): This feature allows for a periodic refresh of SIP sessions through a re-INVITE or UPDATE request. The refresh allows both user agents and proxies to determine whether the SIP session is still active. Two header fields can be defined: Session-Expires, which conveys the lifetime of the session, and Min-SE, which conveys the minimum allowed value for the session timer.
For more information, refer to the “SIP Session Timer Support” section in the Cisco Unified Border Element SIP Support Configuration Guide.
SIP Media Inactivity Detection: The SIP Media Inactivity Detection Timer feature enables Cisco gateways to monitor and disconnect VoIP calls if no Real-Time Control Protocol (RTCP) packets are received within a configurable time period.
For more information, refer to the SIP Media Inactivity Timer section.
The SIP Voice Gateways feature is supported for analog endpoints that are connected to Foreign Exchange Station (FXS) ports or a Cisco VG224 Analog Phone Gateway and controlled by a Cisco call-control system, such as a Cisco Unified Communications Manager (Cisco Unified CM) or a Cisco Unified Communications Manager Express (Cisco Unified CME).
For more information on SIP Gateway features and information about configuring the SIP voice gateway for VoIPv6, see the c_Configuring_a_SIP_Voice_Gateway_for_IPv6_1058563.xml#con_1058563.
SIP is a simple, ASCII-based protocol that uses requests and responses to establish communication among the various components in the network and to ultimately establish a conference between two or more endpoints.
For further information about this feature and information about configuring the SIP voice gateway for VoIPv6, see the Configuring VoIP for IPv6.
SIP is a simple, ASCII-based protocol that uses requests and responses to establish communication among the various components in the network and to ultimately establish a conference between two or more endpoints.
Users in a SIP network are identified by unique SIP addresses. A SIP address is similar to an e-mail address and is in the format of sip:userID@gateway.com. The user ID can be either a username or an E.164 address. The gateway can be either a domain (with or without a hostname) or a specific Internet IPv4 or IPv6 address.
A SIP trunk can operate in one of three modes: SIP trunk in IPv4-only mode, SIP trunk in IPv6-only mode, and SIP trunk in dual-stack mode, which supports both IPv4 and IPv6.
A SIP trunk uses the Alternative Network Address Transport (ANAT) mechanism to exchange multiple IPv4 and IPv6 media addresses for the endpoints in a session. ANAT is automatically enabled on SIP trunks in dual-stack mode. The ANAT Session Description Protocol (SDP) grouping framework allows user agents (UAs) to include both IPv4 and IPv6 addresses in their SDP session descriptions. The UA is then able to use any of its media addresses to establish a media session with a remote UA.
A Cisco Unified Border Element can interoperate between H.323/SIP IPv4 and SIP IPv6 networks in media flow-through mode. In media flow-through mode, both signaling and media flows through the Cisco Unified Border Element, and the Cisco Unified Border Element performs both signaling and media interoperation between H.323/SIP IPv4 and SIP IPv6 networks (see the figure below).
1.
enable
2.
configure
terminal
3.
voice
service
voip
4.
shutdown
[
forced]
1.
enable
2.
configure
terminal
3.
voice
service
voip
4.
sip
5.
call
service
stop
[forced]
SIP service should be shut down before configuring the protocol mode. After configuring the protocol mode as IPv6, IPv4, or dual-stack, SIP service should be reenabled.
1.
enable
2.
configure
terminal
3.
sip-ua
4.
protocol
mode
ipv4
|
ipv6 |
dual-stack [preference {ipv4 |
ipv6}]}
This example shows how to configure the SIP trunk to use dual-stack mode, with IPv6 as the preferred mode. The SIP service must be shut down before any changes are made to protocol mode configuration.
Device(config)# sip-ua
Device(config-sip-ua)# protocol mode dual-stack preference ipv6
ANAT is automatically enabled on SIP trunks in dual-stack mode. Perform this task to disable ANAT in order to use a single-stack mode.
1.
enable
2.
configure
terminal
3.
voice
service
voip
4.
sip
5.
no
anat
Users can configure the source IPv4 or IPv6 address of signaling and media packets to a specific interface’s IPv4 or IPv6 address. Thus, the address that goes out on the packet is bound to the IPv4 or IPv6 address of the interface specified with the bind command.
The bind command also can be configured with one IPv6 address to force the gateway to use the configured address when the bind interface has multiple IPv6 addresses. The bind interface should have both IPv4 and IPv6 addresses to send out ANAT.
When you do not specify a bind address or if the interface is down, the IP layer still provides the best local address.
1.
enable
2.
configure
terminal
3.
voice
service
voip
4.
sip
5.
bind
{control |
media |
all}
source
interface
interface-id [ipv6-address
ipv6-address]
Device(config)# voice service voip Device(config-voi-serv)# sip Device(config-serv-sip)# bind control source-interface fastEthernet 0/0
1.
enable
2.
configure
terminal
3.
sip-ua
4. sip-server {dns: host-name] | ipv4: ipv4–address | ipv6: [ipv6-address] :[port-nums]}
5.
keepalive
target
{{ipv4
:
address |
ipv6
:
address}[:
port] |
dns
:
hostname}
[
tcp
[tls]]
|
udp] [secondary]
Device(config)# sip-ua Device(config-sip-ua)# sip-server ipv6: 2001:DB8:0:0:8:800:200C:417A
1.
enable
2.
configure
terminal
3.
dial-peer
voice
tag
{mmoip |
pots |
vofr |
voip}
4.
destination
pattern
[+
string
T
5.
session
target
{ipv4:
destination-address|
ipv6:
[
destination-address
]|
dns
:
$s$. |
$d$. |
$e$. |
$u$.]
host-name |
enum:table -num |
loopback:rtp |
ras|
sip-server} [:
port
Device(config)# dial-peer voice 29 voip Device(config-dial-peer)# destination-pattern 7777 Device(config-dial-peer)# session target ipv6:2001:DB8:0:0:8:800:200C:417A
1.
enable
2.
configure
terminal
3.
sip-ua
4.
registrar
{dns:
address
|
ipv4:
destination-address [:
port] |
ipv6:
destination-address
:
port] }
aor-domain
expires
seconds [tcp
tls] ]
type [secondary] [scheme
string]
5.
retry
register retries
6.
timers
register milliseconds
Device(config)# sip-ua Device(config-sip-ua)# registrar ipv6: 2001:DB8:0:0:8:800:200C:417A expires 3600 secondary Device(config-sip-ua)# retry register 10 Device((config-sip-ua)# timers register 500
1.
enable
2.
configure
terminal
3.
voice
service
voip
4.
sip
5.
outbound-proxy
{ipv4:
ipv4-address |
ipv6:
ipv6-address |
dns:
host
:
domain} [:
port-number]
To verify the status of SIP Gateway, use the following commands
1.
show
sip-ua
calls
2.
show
sip-ua
connections
3.
show
sip-ua
status
An organization with an IPv4 network can deploy a Cisco UBE on the boundary to connect with the service provider’s IPv6 network (see the figure below).
A Cisco UBE can interoperate between H.323/SIP IPv4 and SIP IPv6 networks in media flow-through mode. In media flow-through mode, both signaling and media flows through the Cisco UBE, and the Cisco UBE performs both signaling and media interoperation between H.323/SIP IPv4 and SIP IPv6 networks (see the figure below).
The Cisco UBE feature adds IPv6 capability to existing VoIP features. This feature adds dual-stack support on voice gateways and MTP, IPv6 support for SIP trunks, and SCCP-controlled analog voice gateways. In addition, the SBC functionality of connecting SIP IPv4 or H.323 IPv4 network to a SIP IPv6 network is implemented on an Cisco UBE to facilitate migration from VoIPv4 to VoIPv6.
Cisco UBE must be configured in IPv6-only or dual-stack mode to support IPv6 calls.
![]() Note | A Cisco UBE interoperates between H.323/SIP IPv4 and SIP IPv6 networks only in media flow-through mode. |
1.
enable
2.
configure
terminal
3.
voice
service
voip
4.
allow-connections
from
type
to
to
type
Device(config)# voice service voip Device(config-voi-serv)# allow-connections h323 to sip
To enable all Session Initiation Protocol (SIP)-related debugging, use the debug ccsip all command in privileged EXEC mode.
To trace the execution path through the call control application programming interface (CCAPI), use the debug voip ccapi inout command.
To enable all Session Initiation Protocol (SIP)-related debugging, use the debug ccsip all command.
To trace the execution path through the call control application programming interface (CCAPI), use the debug voip ccapi inout command.
To enable all Session Initiation Protocol (SIP)-related debugging (when the call is active in Pass through mode), use the debug ccsip all command.
To enable all Session Initiation Protocol (SIP)-related debugging, use the debug ccsip all command.
To enable debugging for Real-Time Transport Protocol (RTP) named event packets, use the debug voip rtp command.
To collect debug information only for signaling events, use the debug vpm signal command.
To show all Session Initiation Protocol (SIP) Service Provider Interface (SPI) message tracing, use the debug ccsip messages command.
To verify that media settings are enabled in the media flowthrough and media flow-around feature, use the following commands:
1.
show call active voice brief
2.
show call active voice compact
3.
show voip rtp connections
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 . An account on Cisco.com is not required.Feature Name |
Releases |
Feature Information |
---|---|---|
IPv6 Dual Stack |
Cisco IOS XE Release 3.3S Cisco IOS XE Release 3.8S Cisco IOS XE Release 3.9S |
Adds IPv6 capability to existing VoIP features on the Cisco Unified Border Element (Enterprise). Additionally, the SBC functionality of connecting SIP IPv4 or H.323 IPv4 network to SIP IPv6 network is implemented on a Cisco Unified Border Element to facilitate migration from VoIPv4 to VoIPv6. The following commands were introduced or modified: None |
DSCP-Based QoS Support |
Cisco IOS XE Release 3.9S |
IPv6 supports this feature. |
RTP/RTCP over IPv6 |
Cisco IOS Release XE 3.9S |
RTP stack supports the ability to create IPv6 connections using IPv6 unicast and multicast addresses as well as IPV4 connections. |