The Network-Based Recording Using Cisco UBE feature supports software-based forking for Real-time Transport Protocol (RTP) streams. Media forking provides the ability to create midcall multiple streams (or branches) of audio associated with a single call and then send the streams of data to different destinations. You can enable Network-Based Recording using Cisco Unified Border Element (Cisco UBE) by configuring specific CLI commands on Cisco UBE or through a call agent. Cisco UBE acts as a recording client and MediaSense Session Initiation Protocol (SIP) recorder acts a recording server.
Functionalities of the recording client, Cisco UBE that is present in signaling and media path of the communication session, are as follows:
Acts as a SIP user agent and sets up a recording session (SIP dialog) with the recording server.
Acts as the source of the recorded media and forwards the recorded media to the recording server.
Sends information periodically to a server that helps the recording server associate the call with media streams and identifies the participants of the call. This information sent to the recording server is called metadata.
MediaSense SIP recorder acts as the recording server. A recording server is a SIP user agent that archives media for extended durations. It provides search and retrieval of the archived media. The recording server is a storage place of the recorded session metadata.
The metadata carried in the SIP session between the recording client and the recording server is to:
Carry the communication session data that describes the call.
Send the metadata to the recording server. The recording server uses the metadata to associate communication sessions involving two or more participants with media streams.
The call leg that is created between the recording client and the recording server is known as the recording session.
Prerequisites for Network-Based Recording Using Cisco UBE
You must have an ISR G2 router equipped with the unified communication technology package configured as a Cisco UBE in flow-through mode for the Network-Based Recording Using Cisco UBE feature to function.
Restrictions for Network-Based Recording Using Cisco UBE
The Network-Based Recording Using Cisco UBE feature is supported only for SIP-SIP call flows.
The Network-Based Recording Using Cisco UBE feature is supported only on ISR G2 platforms (2901, 2911, 2921, 2951, 3945, 3945E).
Information About Network-Based Recording Using Cisco UBE
The Open Recording Architecture (ORA) comprises of elements, such as application management server and SIP bridge, to support IP-based recording. The ORA IP enables recording by solving topology issues, which accelerates the adoption of Cisco unified communication solutions.
Following are the three layers of the ORA architecture:
The ORA network layer is comprises call control systems, media sources, and IP foundation components, such as routers and switches.
Capture and Media Processing Layer
The ORA capture and media processing layer includes core functions of ORA--terminating media streams, storage of media and metadata, and speech analytics that can provide real-time events for applications.
Application Layer
The ORA application layer supports in-call and post-call applications through open programming interfaces. In-call applications include applications that make real-time business decisions like whether to record a particular call or not, controls pause and resume from Interactive Voice Response (IVR) or agent desktop systems, and performs metadata tagging and encryption key exchange at the call setup.
Post-call applications include the following:
Traditional compliance search, replay, and quality monitoring.
Advanced capabilities, such as speech analytics, transcription, and phonetic search.
Figure 2
shows the ORA SBC-based media forking with Cisco Unified
CallManager (Cisco UCM) topology for media forking. This topology supports
replication of media packets to allow recording the caller agent. It also
enables Cisco UBE to establish full-duplex communication with the recording
server. In this topology, SIP recording trunk is enhanced to have additional
call metadata.
ORA SBC-Based Media Forking without Cisco UCM
Figure 3
shows the ORA SBC-based media forking without the Cisco UCM
topology. This topology supports static configuration on Cisco UBE and the
replication of media packets to allow recording caller-agent and full-duplex
interactions at an IP call recording server.
SIP Recorder Interface
SIP is used as a protocol between Cisco UBE and the MediaSense SIP server. Extensions are made to SIP to carry the recording session information needed for the recording server. This information carried in SIP sessions between the recording client and the recording server is called metadata.
Metadata is the information that is passed by the recording client to the recording server in a SIP session. Metadata describes the communication session and its media streams.
Metadata is used by the recording server to:
Identify participants of the call.
Associate media streams with the participant information. Each participant can have one or more media streams, such as audio, and video.
Identify the participant change due to transfers during the call.
The recording server uses the metadata information along with other SIP message information, such as dialog ID and time and date header, to derive a unique key. The recording server uses this key to store media streams and associate the participant information with the media streams.
How to Configure Network-Based Recording using Cisco UBE
You can configure and verify the Network-Based Recording using Cisco UBE feature by using one of the following methods:
Exits media class recorder parameter configuration mode.
Configuring the Media Class for a Dial Peer
Perform this task to configure the media class for a dial peer. You must configure a dial peer to connect to MediaSense. This dial peer is matched with Cisco UBE and a call is set up to MediaSense.
Specifies the voice coder rate of speech for a dial peer.
Step 8
end
Example:
Router(config-dial-peer)# end
Exits dial peer configuration mode and returns to privileged EXEC mode.
Verifying the Network-Based Recording Using Cisco UBE Configuration
Perform this task to verify the configuration of the Network-Based Recording Using Cisco UBE Configuration feature. The
show commands can be entered in any order.
Displays active recording Media Service Provider (MSP) session information.
Example:
Router# show voip recmsp session
RECMSP active sessions:
MSP Call-ID AnchorLeg Call-ID ForkedLeg Call-ID
143 141 145
Found 1 active sessions
Step 4
showvoiprecmspsessiondetailcall-idcall-id
Displays detailed information about the recording MSP Call ID.
Example:
Router# show voip recmsp session detail call-id 145
RECMSP active sessions:
Detailed Information
=========================
Recording MSP Leg Details:
Call ID: 143
GUID : 7C5946D38ECD
AnchorLeg Details:
Call ID: 141
Forking Stream type: voice-nearend
Participant: 708090
Non-anchor Leg Details:
Call ID: 140
Forking Stream type: voice-farend
Participant: 10000
Forked Leg Details:
Call ID: 145
Near End Stream CallID 145
Stream State ACTIVE
Far End stream CallID 146
Stream State ACTIVE
Found 1 active sessions
Step 5
showvoiprtpforking
Displays RTP media-forking connections.
Example:
Router# show voip rtp forking
VoIP RTP active forks :
Fork 1
stream type voice-only (0): count 0
stream type voice+dtmf (1): count 0
stream type dtmf-only (2): count 0
stream type voice-nearend (3): count 1
remote ip 9.42.29.7, remote port 38526, local port 18648
codec g711ulaw, logical ssrc 0x53
packets sent 29687, packets received 0
stream type voice+dtmf-nearend (4): count 0
stream type voice-farend (5): count 1
remote ip 9.42.29.7, remote port 50482, local port 17780
codec g711ulaw, logical ssrc 0x55
packets sent 29686, packets received 0
stream type voice+dtmf-farend (6): count 0
stream type video (7): count
Step 6
showcallactivevoicecompact
Displays a compact version of voice calls in progress.
Example:
Router# show call active voice compact
<callID> A/O FAX T<sec> Codec type Peer Address IP R<ip>:<udp>
Total call-legs: 3
140 ANS T644 g711ulaw VOIP P10000 9.42.30.32:18638
141 ORG T644 g711ulaw VOIP P708090 9.42.30.189:26184
145 ORG T643 g711ulaw VOIP P595959 9.42.29.7:38526
Step 7
showsip-uacalls
Displays active user agent client (UAC) and user agent server (UAS) information on SIP calls.
Example:
Router# show sip-ua calls
Total SIP call legs:3, User Agent Client:2, User Agent Server:1
SIP UAC CALL INFO
Call 1
SIP Call ID : 99EA5118-506211E0-80C6E01B-4C27AA62@9.42.30.10
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 10000
Called Number : 708090
Bit Flags : 0xC04018 0x10000100 0x80
CC Call ID : 141
Source IP Address (Sig ): 9.42.30.10
Destn SIP Req Addr:Port : [9.42.30.5]:5060
Destn SIP Resp Addr:Port: [9.42.30.5]:5060
Destination Name : 9.42.30.5
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 141
Stream Type : voice+dtmf (1)
Stream Media Addr Type : 1
Negotiated Codec : g711ulaw (160 bytes)
Codec Payload Type : 0
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [9.42.30.10]:19256
Media Dest IP Addr:Port : [9.42.30.189]:26184
Options-Ping ENABLED:NO ACTIVE:NO
Call 2
SIP Call ID : 9A6D8922-506211E0-80CEE01B-4C27AA62@9.42.30.10
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number :
Called Number : 595959 Recoding server number
Bit Flags : 0xC04018 0x10800100 0x80
CC Call ID : 145
Source IP Address (Sig ): 9.42.30.10
Destn SIP Req Addr:Port : [9.42.29.7]:5060
Destn SIP Resp Addr:Port: [9.42.29.7]:5060
Destination Name : 9.42.29.7
Number of Media Streams : 2
Number of Active Streams: 2
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 145
Stream Type : voice-nearend (3)
Stream Media Addr Type : 1
Negotiated Codec : g711ulaw (160 bytes)
Codec Payload Type : 0
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [9.42.30.10]:18648
Media Dest IP Addr:Port : [9.42.29.7]:38526
Media Stream 2
State of the stream : STREAM_ACTIVE
Stream Call ID : 146
Stream Type : voice-farend (5)
Stream Media Addr Type : 1
Negotiated Codec : g711ulaw (160 bytes)
Codec Payload Type : 0
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [9.42.30.10]:17780
Media Dest IP Addr:Port : [9.42.29.7]:50482
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Client(UAC) calls: 2
SIP UAS CALL INFO
Call 1
SIP Call ID : 7CF44DF3-506611E0-8ED2B9D4-CA68C314@9.42.30.32
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 10000
Called Number : 708090
Bit Flags : 0x8C4401C 0x10000100 0x4
CC Call ID : 140
Source IP Address (Sig ): 9.42.30.10
Destn SIP Req Addr:Port : [9.42.30.32]:5060
Destn SIP Resp Addr:Port: [9.42.30.32]:52757
Destination Name : 9.42.30.32
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 140
Stream Type : voice+dtmf (0)
Stream Media Addr Type : 1
Negotiated Codec : g711ulaw (160 bytes)
Codec Payload Type : 0
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [9.42.30.10]:18792
Media Dest IP Addr:Port : [9.42.30.32]:18638
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Server(UAS) calls: 1
Configuration Examples for Network-Based Recording using Cisco UBE
No new or modified RFCs are supported by this feature, and support for existing RFCs has not been modified by this feature.
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Feature Information for Network-Based Recording using Cisco UBE
GUID-B63FA45E-FB93-4612-B5D3-73639725826DC lists the release history for this feature.
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GUID-B63FA45E-FB93-4612-B5D3-73639725826DC 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.
Table 1
Feature Information for Network-Based Recording using Cisco UBE
Feature Name
Releases
Feature Information
Network-Based Recording Using Cisco UBE
15.2(1)T
The Network-Based Recording Using Cisco UBE feature supports software-based forking for RTP streams.
The following commands were introduced or modified:
mediaclass,
mediaprofilerecorder,
media-recording,
recorderparameter,
recorderprofile,
showvoiprecmspsession.