Cisco Unified Border Element (Enterprise) Protocol-Independent Features and Setup Configuration Guide, Cisco IOS XE Release 3S
Network-Based Recording
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Network-Based Recording

Network-Based Recording

The Network-Based Recording 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 and video associated with a single call and then send the streams of data to different destinations. To enable network-based recording using Cisco Unified Border Element (CUBE), you can configure specific commands or use a call agent. CUBE acts as a recording client and MediaSense Session Initiation Protocol (SIP) recorder acts a recording server.

Feature Information for Network-Based Recording

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 software image support. Cisco Feature Navigator enables you to determine which software images support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to http:/​/​www.cisco.com/​go/​cfn. An account on Cisco.com is not required.

Table 1 Feature Information for Network-Based Recording

Feature Name

Releases

Feature Information

Audio-only Stream Forking of Video Call

Cisco IOS 15.4(3)M

Cisco IOS XE 3.13S

The Audio-only Stream Forking of Video Call feature supports CUBE-based forking and recording of only audio calls in a call that includes both audio and video. The following commands were introduced: media-type audio.

Network-Based Recording of Video Calls Using CUBE

Cisco IOS 15.3(3)M

Cisco IOS XE 3.10S

The Network-Based Recording of Video Calls using CUBE feature supports forking and recording of video calls.

Network-Based Recording of Audio Calls Using CUBE

Cisco IOS 15.2(1)T

Cisco IOS XE 3.8S

The Network-Based Recording of Audio Calls using CUBE feature supports forking for RTP streams.

The following commands were introduced or modified: media class, media profile recorder, media-recording, recorder parameter, recorder profile, show voip recmsp session.

Restrictions for Network-Based Recording

  • Network-based recording is not supported for the following calls:

    • Calls that are not Session Initiation Protocol (SIP) SIP-to-SIP

    • Flow-around calls

    • Session Description Protocol (SDP) pass-through calls

    • Real-time Transport Protocol (RTP) loopback calls

    • High-density transcoder calls

    • IPv6-to-IPv6 calls

    • IPv6-to-IPv4 calls with IPv4 endpoint.

    • Secure Real-time Transport Protocol (SRTP) passthrough calls

    • SRTP-RTP calls with forking for SRTP leg (forking is supported for the RTP leg)

    • Resource Reservation Protocol (RSVP)

    • Multicast music on hold (MOH)

  • Any media service parameter change via Re-INVITE or UPDATE from Recording server is not supported Midcall renegotiation and supplementary services can be done through the primary call only.

  • Media service parameter change via Re-INVITE or UPDATE message from the recording server is not supported

  • Recording is not supported if CUBE is running a TCL IVR application.

  • Media mixing on forked streams is not supported

Restrictions for Video Recording

  • If the main call has multiple video streams (m-lines), the video streams other than the first video m-line are not forked.

  • Application media streams of the primary call are not forked to the recording server.

  • Forking is not supported if the anchor leg or recording server is on IPv6.

  • High availability is not supported on forked video calls.

Information About Network-Based Recording Using CUBE

Deployment Scenarios for CUBE-based Recording

CUBE as a recording client has the following functions:

  • 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 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.

Given below is a typical deployment scenario of a CUBE-based recording solution. The information flow is described below:

Figure 1. Deployment Scenario for CUBE-based Recording Solution



  1. Incoming call from SIP trunk.

  2. Outbound call to a Contact Centre

  3. Media between endpoints flowthrough CUBE

  4. CUBE sets up a new SIP session with MediaSense based on policy.

  5. CUBE forks RTP media to MediaSense. For an audio call, audio is forked. For a video call, both audio and video are .forked. For an audio-only configuration in a audio-video call, only audio is forked. There will be two or four m-lines to the recording server, based on the type of recording

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.

Open Recording Architecture

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:

Network Layer

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 (for example, whether to record a particular call or not), control pause and resume from Interactive Voice Response (IVR) or agent desktop systems, and perform 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.

  • Custom enterprise integration.

  • Enterprise-wide policy management.

Media Forking Topologies

The following topologies support media forking:

Media Forking with Cisco UCM

The figure below illustrates media forking with Cisco Unified CallManager (Cisco UCM) topology. This topology supports replication of media packets to allow recording by the caller agent. It also enables CUBE to establish full-duplex communication with the recording server. In this topology, SIP recording trunk is enhanced to have additional call metadata.



Media Forking without Cisco UCM

The topology below shows media forking without the Cisco UCM topology. This topology supports static configuration on CUBE 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 CUBE 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

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

Configuring Network-Based Recording (with Media Profile Recorder)

SUMMARY STEPS

    1.    enable

    2.    configure terminal

    3.    media profile recorder profile-tag

    4.    (Optional) media-type audio

    5.    media-recording dial-peer-tag [dial-peer-tag2...dial-peer-tag5]

    6.    exit

    7.    media class tag

    8.    recorder profile tag

    9.    exit

    10.    dial-peer voice dummy-recorder-dial-peer-tag voip

    11.    media-class tag

    12.    destination-pattern [+] string [T]

    13.    session protocol sipv2

    14.    session target ipv4:[recording-server-destination-address | recording-server-dns]

    15.    session transport tcp

    16.    end


DETAILED STEPS
     Command or ActionPurpose
    Step 1 enable


    Example:
    Device> enable
     

    Enables privileged EXEC mode.

    • Enter your password if prompted.

     
    Step 2 configure terminal


    Example:
    Device# configure terminal
     

    Enters global configuration mode.

     
    Step 3 media profile recorder profile-tag


    Example:
    Device(config)# media profile recorder 100
     

    Configures the media profile recorder and enters media profile configuration mode.

     
    Step 4 media-type audio


    Example:
    Device(cfg-mediaprofile)# media-type audio
     
    (Optional)

    Configures recording of audio only in a call with both audio and video. If this configuration is not done, both audio and video are recorded.

     
    Step 5 media-recording dial-peer-tag [dial-peer-tag2...dial-peer-tag5]


    Example:
    Device(cfg-mediaprofile)# media-recording 8000 8001 8002
     

    Configures the dial-peers that need to be configured

    Note   

    You can specify a maximum of five dial-peer tags.

     
    Step 6 exit


    Example:
    Device(cfg-mediaprofile)# exit
     

    Exits media profile configuration mode.

     
    Step 7 media class tag


    Example:
    Device(config)# media class 100
     

    Configures a media class and enters media class configuration mode.

     
    Step 8 recorder profile tag


    Example:
    Device(cfg-mediaclass)# recorder profile 100
     

    Configures the media profile recorder.

     
    Step 9 exit


    Example:
    Device(cfg-mediaclass)# exit
     

    Exits media class configuration mode.

     
    Step 10 dial-peer voice dummy-recorder-dial-peer-tag voip


    Example:
    Device(config)# dial-peer voice 8000 voip
     

    Configures a recorder dial peer and enters dial peer voice configuration mode.

     
    Step 11 media-class tag


    Example:
    Device(config-dial-peer)# media-class 100
     

    Configures media class on a dial peer.

     
    Step 12 destination-pattern [+] string [T]


    Example:
    Device(config-dial-peer)# destination-pattern 595959
     

    Specifies either the prefix or the full E.164 telephone number (depending on your dial plan) to be used for a dial peer.

     
    Step 13 session protocol sipv2


    Example:
    Device(config-dial-peer)# session protocol sipv2
     

    Configures the VoIP dial peer to use Session Initiation Protocol (SIP).

     
    Step 14 session target ipv4:[recording-server-destination-address | recording-server-dns]


    Example:
    Device(config-dial-peer)# session target ipv4:10.42.29.7
     

    Specifies a network-specific address for a dial peer. Keyword and argument are as follows:

    • ipv4: destination address --IP address of the dial peer, in this format: xxx.xxx.xxx.xxx

     
    Step 15session transport tcp


    Example:
    Device(config-dial-peer)# session transport tcp
     

    Configures a VoIP dial peer to use Transmission Control Protocol (TCP).

     
    Step 16 end


    Example:
    Device(config-dial-peer)# end
     

    Returns to privileged EXEC mode.

     

    Configuring Network-Based Recording (without Media Profile Recorder)

    SUMMARY STEPS

      1.    enable

      2.    configure terminal

      3.    media class tag

      4.    recorder parameter

      5.    (Optional) media-type audio

      6.    media-recording dial-peer-tag

      7.    exit

      8.    exit

      9.    dial-peer voice dummy-recorder-dial-peer-tag voip

      10.    media-class tag

      11.    destination-pattern [+] string [T]

      12.    session protocol sipv2

      13.    session target ipv4:[recording-server-destination-address | recording-server-dns]

      14.    session transport tcp

      15.    end


    DETAILED STEPS
       Command or ActionPurpose
      Step 1 enable


      Example:
      Device> enable
       

      Enables privileged EXEC mode.

      • Enter your password if prompted.

       
      Step 2 configure terminal


      Example:
      Device# configure terminal
       

      Enters global configuration mode.

       
      Step 3 media class tag


      Example:
      Device(config)# media class 100
       

      Configures the media class and enters media class configuration mode.

       
      Step 4 recorder parameter


      Example:
      Device(cfg-mediaclass)# recorder parameter
       

      Enters media class recorder parameter configuration mode to enable you to configure recorder-specific parameters.

       
      Step 5 media-type audio


      Example:
      Device(cfg-mediaprofile)# media-type audio
       
      (Optional)

      Configures recording of audio only in a call with both audio and video.

      Note   

      If this configuration is not done, both audio and video are recorded.

       
      Step 6 media-recording dial-peer-tag


      Example:
      Device(cfg-mediaclass-recorder)# media-recording 8000, 8001, 8002
       

      Configures voice-class recording parameters.

      Note   

      You can specify a maximum of five dial-peer tags.

       
      Step 7 exit


      Example:
      Device(cfg-mediaclass-recorder)# exit
       

      Exits media class recorder parameter configuration mode.

       
      Step 8 exit


      Example:
      Device(cfg-mediaclass)# exit
       

      Exits media class configuration mode.

       
      Step 9 dial-peer voice dummy-recorder-dial-peer-tag voip


      Example:
      Device(config)# dial-peer voice 8000 voip
       

      Configures a recorder dial peer and enters dial peer voice configuration mode.

       
      Step 10 media-class tag


      Example:
      Device(config-dial-peer)# media-class 100
       

      Configures media class on a dial peer.

       
      Step 11 destination-pattern [+] string [T]


      Example:
      Device(config-dial-peer)# destination-pattern 595959
       

      Specifies either the prefix or the full E.164 telephone number (depending on your dial plan) to be used for a dial peer. Keywords and arguments are as follows:

       
      Step 12 session protocol sipv2


      Example:
      Device(config-dial-peer)# session protocol sipv2
       

      Configures the VoIP dial peer to use Session Initiation Protocol (SIP).

       
      Step 13 session target ipv4:[recording-server-destination-address | recording-server-dns]


      Example:
      Device(config-dial-peer)# session target ipv4:10.42.29.7
       

      Specifies a network-specific address for a dial peer. Keyword and argument are as follows:

      • ipv4: destination address --IP address of the dial peer, in this format: xxx.xxx.xxx.xxx

       
      Step 14session transport tcp


      Example:
      Device(config-dial-peer)# session transport tcp
       

      Configures a VoIP dial peer to use Transmission Control Protocol (TCP).

       
      Step 15 end


      Example:
      Device(config-dial-peer)# end
       

      Returns to privileged EXEC mode.

       

      Verifying the Network-Based Recording Using CUBE

      Perform this task to verify the configuration of the Network-Based Recording Using CUBE. The show and debug commands can be entered in any order.

      SUMMARY STEPS

        1.    enable

        2.    show voip rtp connections

        3.    show voip recmsp session

        4.    show voip recmsp session detail call-id call-id

        5.    show voip rtp forking

        6.    show call active voice compact

        7.    show call active video compact

        8.    show sip-ua calls

        9.    show call active video brief

        10.    debug ccsip messages (for audio calls)

        11.    debug ccsip messages (for video calls)

        12.    debug ccsip messages (for audio-only recording in a call with both audio and video)

        13.    Enter one of the following:

        • debug ccsip all
        • debug voip recmsp all
        • debug voip ccapi all
        • debug voip fpi all (for ASR devices only)


      DETAILED STEPS
        Step 1   enable

        Enables privileged EXEC mode.



        Example:
        Device> enable
        
        Step 2   show voip rtp connections

        Displays Real-Time Transport Protocol (RTP) connections. Two extra connections are displayed for forked legs.



        Example:
        Device# show voip rtp connections
        
        VoIP RTP Port Usage Information:
        Max Ports Available: 8091, Ports Reserved: 101, Ports in Use: 8
        Port range not configured, Min: 16384, Max: 32767
        
                                                        Ports       Ports       Ports
        Media-Address Range                             Available   Reserved    In-use
        
        Default Address-Range                           8091        101         8
        
        VoIP RTP active connections :
        No. CallId     dstCallId  LocalRTP RmtRTP LocalIP                                       RemoteIP                           
        1     1          2          16384    20918  10.104.45.191                               10.104.8.94                        
        2     2          1          16386    17412  10.104.45.191                               10.104.8.98                        
        3     3          4          16388    29652  10.104.45.191                               10.104.8.98                        
        4     4          3          16390    20036  10.104.45.191                               10.104.8.94                        
        5     6          5          16392    58368  10.104.45.191                               10.104.105.232                     
        6     7          5          16394    53828  10.104.45.191                               10.104.105.232                     
        7     8          5          16396    39318  10.104.45.191                               10.104.105.232                     
        8     9          5          16398    41114  10.104.45.191                               10.104.105.232                     
        Found 8 active RTP connections
        
        Step 3   show voip recmsp session

        Displays active recording Media Service Provider (MSP) session information internal to CUBE.



        Example:
        Device# 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   show voip recmsp session detail call-id call-id

        Displays detailed information about the recording MSP Call ID.



        Example:
        Device# 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
        
        Device# show voip recmsp session detail call-id 5
        
        RECMSP active sessions:
        Detailed Information
        =========================
        Recording MSP Leg Details:
        Call ID: 5
        GUID : 1E01B6000000
        
        AnchorLeg Details:
        Call ID: 1
        Forking Stream type: voice-nearend
        Forking Stream type: video-nearend
        Participant: 1777
        
        Non-anchor Leg Details:
        Call ID: 2
        Forking Stream type: voice-farend
        Forking Stream type: video-farend
        Participant: 1888
        
        Forked Leg Details:
        Call ID: 6
        Voice Near End Stream CallID 6
        Stream State ACTIVE
        Voice Far End stream CallID 7
        Stream State ACTIVE
        Video Near End stream CallID 8
        Stream State ACTIVE
        Video Far End stream CallID 9
        Stream State ACTIVE
        Found 1 active sessions
        
        

        Output Field

        Description

        Stream State Displays the state of the call. This can be ACTIVE or HOLD.
        Msp Call-Id

        Displays an internal Media service provider call ID and forking related statistics for an active forked call.

        Anchor Leg Call-id

        Displays an internal anchor leg ID, which is the dial peer where forking enabled. The output displays the participant number and stream type. Stream type voice-near end indicates the called party side.

        Non-Anchor Call-id

        Displays an internal non-anchor leg ID, which is the dial peer where forking is not enabled. The output displays the participant number and stream type. Stream type voice-near end indicates the called party side.

        Forked Call-id

        This forking leg call-id will show near-end and far-end stream call-id details with state of the Stream .

        Displays an internal foked leg ID. The output displays near-end and far-end details of a stream.

        Step 5   show voip rtp forking

        Displays RTP media-forking connections.



        Example:
        Device# 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 10.42.29.7,  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 10.42.29.7,  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

        Output Field

        Description

        remote ip 10.42.29.7, remote port 38526, local port 18648

        Recording server IP, recording server port, and local CUBE device port where data for stream 1 was first sent from.

        remote ip 10.42.29.7, remote port 50482, local port 17780

        Recording server IP, recording server port, and local CUBE device port where data for stream 2 was first sent from.

        packets sent 29686

        Number of packets sent to the recorder

        codec g711ulaw Codec negotiated for the recording leg.
        Step 6   show call active voice compact

        Displays a compact version of voice calls in progress. An additional call leg is displayed for media forking.



        Example:
        Device# 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       10.42.30.32:18638
               141 ORG     T644   g711ulaw    VOIP        P708090      10.42.30.189:26184
               145 ORG     T643   g711ulaw    VOIP        P595959      10.42.29.7:38526
        
        Step 7   show call active video compact

        Displays a compact version of video calls in progress.



        Example:
        Device# show call active video compact
        
        <callID>  A/O FAX T<sec> Codec       type        Peer Address       IP R<ip>:<udp>
        Total call-legs: 3
                 1 ANS     T14    H264        VOIP-VIDEO  P1777      10.104.8.94:20036
                 2 ORG     T14    H264        VOIP-VIDEO  P1888      10.104.8.98:29652
                 6 ORG     T13    H264        VOIP-VIDEO  P1234   10.104.105.232:39318
        
        Step 8   show sip-ua calls

        Displays active user agent client (UAC) and user agent server (UAS) information on SIP calls.



        Example:
        Device# 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@10.42.30.10
           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 ): 10.42.30.10
           Destn SIP Req Addr:Port : [10.42.30.5]:5060
           Destn SIP Resp Addr:Port: [10.42.30.5]:5060
           Destination Name        : 10.42.30.5
           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: [10.42.30.10]:19256
             Media Dest IP Addr:Port  : [10.42.30.189]:26184
        Options-Ping    ENABLED:NO    ACTIVE:NO
        Call 2
        SIP Call ID                : 9A6D8922-506211E0-80CEE01B-4C27AA62@10.42.30.10
           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 ): 10.42.30.10
           Destn SIP Req Addr:Port : [10.42.29.7]:5060
           Destn SIP Resp Addr:Port: [10.42.29.7]:5060
           Destination Name        : 10.42.29.7
           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: [10.42.30.10]:18648
             Media Dest IP Addr:Port  : [10.42.29.7]: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: [10.42.30.10]:17780
             Media Dest IP Addr:Port  : [10.42.29.7]: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@10.42.30.32
           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 ): 10.42.30.10
           Destn SIP Req Addr:Port : [10.42.30.32]:5060
           Destn SIP Resp Addr:Port: [10.42.30.32]:52757
           Destination Name        : 10.42.30.32
           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: [10.42.30.10]:18792
             Media Dest IP Addr:Port  : [10.42.30.32]:18638
        Options-Ping    ENABLED:NO    ACTIVE:NO
           Number of SIP User Agent Server(UAS) calls: 1
        Step 9   show call active video brief

        Displays a truncated version of video calls in progress.



        Example:
        Device# show call active video brief
        
        Telephony call-legs: 0
        SIP call-legs: 3
        H323 call-legs: 0
        Call agent controlled call-legs: 0
        SCCP call-legs: 0
        Multicast call-legs: 0
        Total call-legs: 3
        
        0    : 1 87424920ms.1 (*12:23:53.573 IST Wed Jul 17 2013) +1050 pid:1 Answer 1777 active
         dur 00:00:46 tx:5250/1857831 rx:5293/1930598 dscp:0 media:0 audio tos:0xB8 video tos:0x88
         IP 10.104.8.94:20036 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms H264 TextRelay: off Transcoded: No
        … 
        0    : 2 87424930ms.1 (*12:23:53.583 IST Wed Jul 17 2013) +1040 pid:2 Originate 1888 active
         dur 00:00:46 tx:5293/1930598 rx:5250/1857831 dscp:0 media:0 audio tos:0xB8 video tos:0x88
         IP 10.104.8.98:29652 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms H264 TextRelay: off Transcoded: No
        …
        0    : 6 87425990ms.1 (*12:23:54.643 IST Wed Jul 17 2013) +680 pid:1234 Originate 1234 active
         dur 00:00:46 tx:10398/3732871 rx:0/0 dscp:0 media:0 audio tos:0xB8 video tos:0x0
         IP 10.104.105.232:39318 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms H264 TextRelay: off Transcoded: No
        … 
        
        
        Step 10   debug ccsip messages (for audio calls)
        Sent:
        INVITE sip:22222@10.42.29.7:5060 SIP/2.0
        Via: SIP/2.0/TCP 10.42.30.10:5060;branch=z9hG4bKB622CF
        X-Cisco-Recording-Participant: sip:708090@10.42.30.5;media-index="0"
        X-Cisco-Recording-Participant: sip:10000@10.42.30.32;media-index="1"
        From: <sip:10.42.30.10>;tag=5096700-1E1A
        To: <sip:595959@10.42.29.7>
        Date: Fri, 18 Mar 2011 07:01:50 GMT
        Call-ID: 6E6CF813-506411E0-80EAE01B-4C27AA62@10.42.30.10
        Supported: 100rel,timer,resource-priority,replaces,sdp-anat
        Min-SE:  1800
        Cisco-Guid: 1334370502-1348997600-2396699092-3395863316
        User-Agent: Cisco-SIPGateway/IOS-15.2(0.0.2)PIA16
        Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
        CSeq: 101 INVITE
        Max-Forwards: 70
        Timestamp: 1300431710
        Contact: <sip:10.42.30.10:5060;transport=tcp>
        Expires: 180
        Allow-Events: telephone-event
        Content-Type: application/sdp
        Content-Disposition: session;handling=required
        Content-Length: 449
        v=0
        o=CiscoSystemsSIP-GW-UserAgent 3021 3526 IN IP4 10.42.30.10
        s=SIP Call
        c=IN IP4 10.42.30.10
        t=0 0
        m=audio 24544 RTP/AVP 0 101 19
        c=IN IP4 10.42.30.10
        a=rtpmap:0 PCMU/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=rtpmap:19 CN/8000
        a=ptime:20
        a=sendonly
        m=audio 31166 RTP/AVP 0 101 19
        c=IN IP4 10.42.30.10
        a=rtpmap:0 PCMU/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=rtpmap:19 CN/8000
        a=ptime:20
        a=sendonly
        Received:
        SIP/2.0 200 Ok
        Via: SIP/2.0/TCP 10.104.46.198:5060;branch=z9hG4bK13262B
        To: <sip:23232323@10.104.46.201>;tag=ds457251f
        From: <sip:10.104.46.198>;tag=110B66-1CBC
        Call-ID: 7142FB-9A5011E0-801EF71A-59B4D258@10.104.46.198
        CSeq: 101 INVITE
        Content-Length: 206
        Contact: <sip:23232323@10.104.46.201:5060;transport=tcp>
        Content-Type: application/sdp
        Allow: INVITE, BYE, CANCEL, ACK, NOTIFY, INFO, UPDATE
        Server: Cisco-ORA/8.5
        v=0
        o=CiscoORA 2187 1 IN IP4 10.104.46.201
        s=SIP Call
        c=IN IP4 10.104.46.201
        t=0 0
        m=audio 54100 RTP/AVP 0
        a=rtpmap:0 PCMU/8000
        a=recvonly
        m=audio 39674 RTP/AVP 0
        a=rtpmap:0 PCMU/8000
        a=recvonly
        
        Sent:
        ACK sip:23232323@10.104.46.201:5060;transport=tcp SIP/2.0
        Via: SIP/2.0/TCP 10.104.46.198:5060;branch=z9hG4bK141B87
        From: <sip:10.104.46.198>;tag=110B66-1CBC
        To: <sip:23232323@10.104.46.201>;tag=ds457251f
        Date: Mon, 20 Jun 2011 08:42:01 GMT
        Call-ID: 7142FB-9A5011E0-801EF71A-59B4D258@10.104.46.198
        Max-Forwards: 70
        CSeq: 101 ACK
        Allow-Events: telephone-event
        Content-Length: 0
        

        Output Field

        Description

        INVITE sip:22222@10.42.29.7:5060 SIP/2.0

        22222 is the destination pattern or the number of recording server and is configured under the recorder dial peer.

        X-Cisco-Recording-Participant: sip:708090@10.42.30.5;media-index="0"

        Cisco proprietary header with originating and terminating participant number and IP address used to communicate to the recording server

        Cisco-Guid: 1334370502-1348997600-2396699092-3395863316

        GUID is the same for the primary call and forked call .

        m=audio 24544 RTP/AVP 0 101 19

        First m-line of participant with payload type and codec information .

        m=audio 31166 RTP/AVP 0 101 19

        Second m- line of another participant with codec info and payload type.

        a=sendonly

        CUBE is always in send only mode towards Recording server.

        a=recvonly

        Recording server is in receive mode only.

        Step 11   debug ccsip messages (for video calls)
        Sent: INVITE sip:575757@9.45.38.39:7686 SIP/2.0
        
        .
        .
        Via: SIP/2.0/UDP 9.41.36.41:5060;branch=z9hG4bK2CC2408
        X-Cisco-Recording-Participant: sip:1777@10.104.45.207;media-
        index="0 2“
        X-Cisco-Recording-Participant: sip:1888@10.104.45.207;media-   index="1 3“
        .
        .
        Cisco-Guid: 0884935168-0000065536-0000000401-3475859466
        .
        .
        v=0
        .
        .
        .
        m=audio 17232 RTP/AVP 0 19
        .
        .
        a=sendonly
        m=audio 17234 RTP/AVP 0 19
        .
        .
        a=sendonly
        
        m=video 17236 RTP/AVP 126
        .
        .
        .
        
        
        a=fmtp:126 profile-level-id=42801E;packetization-mode=1
        a=sendonly
        m=video 17238 RTP/AVP 126
        .
        .
        
        .
        a=fmtp:126 profile-level-id=42801E;packetization-mode=1
        a=sendonly
        
        

        Output Field

        Description

        Sent: INVITE sip:575757@9.45.38.39:7686 SIP/2.0

        22222 is the destination pattern or the number of recording server and is configured under the recorder dial peer.

        X-Cisco-Recording-Participant: sip:1777@10.104.45.207;media- index="0 2“ X-Cisco-Recording-Participant: sip:1888@10.104.45.207;media- index="1 3“ Cisco proprietary header with originating and terminating participant number and IP address used to communicate to the recording server

        Cisco-Guid: 0884935168-0000065536-0000000401-3475859466

        GUID is the same for the primary call and forked call .

        m=audio 17232 RTP/AVP 0 19

        First m-line of participant with payload type and audio codec.

        m=audio 17234 RTP/AVP 0 19

        Second m-line of another participant with payload type and audio codec.

        m=video 17236 RTP/AVP 126

        Third m-line of participant with video payload type and codec info .

        m=video 17238 RTP/AVP 126

        Fourth m-line of another participant with video payload type and codec info .

        a=sendonly

        CUBE is always in send only mode towards Recording server.

        Receive:
        SIP/2.0 200 OK
        .
        .
        .
        
        v=0
        .
        .
        m=audio 1592 RTP/AVP 0
        .
        .
        a=recvonly
        m=audio 1594 RTP/AVP 0
        .
        .
        a=recvonly
        m=video 1596 RTP/AVP 126
        .
        .
        a=fmtp:97 profile-level-id=420015
        a=recvonly
        m=video 1598 RTP/AVP 126
        .
        .
        a=fmtp:126 profile-level-id=420015
        a=recvonly
        Sent: 
        ACK sip:9.45.38.39:7686;transport=UDP SIP/2.0
        
        Via: SIP/2.0/UDP 9.41.36.41:5060;branch=z9hG4bK2CD7
        
        From: <sip:9.41.36.41>;tag=1ECFD128-24DF
        
        To: <sip:575757@9.45.38.39>;tag=16104SIPpTag011
        
        Date: Tue, 19 Mar 2013 11:40:01 GMT
        
        Call-ID: FFFFFFFF91E00FE6-FFFFFFFF8FC011E2-FFFFFFFF824DF469-FFFFFFFFB6661C06@9.41.36.41
        
        Max-Forwards: 70
        
        CSeq: 101 ACK
        
        Allow-Events: telephone-event
        
        Content-Length: 0
        

        Output Field

        Description

        m=audio 1592 RTP/AVP 0

        First m-line of recording server after it started listening.

        m=audio 1594 RTP/AVP 0

        Second m-line of recording server after it started listening.

        m=video 1596 RTP/AVP 126

        Third m-line of recording server after it started listening.

        m=video 1598 RTP/AVP 126

        Fourth m-line of recording server after it started listening.

        a=recvonly

        Recording server in receive only mode.

        Step 12   debug ccsip messages (for audio-only recording in a call with both audio and video)

        Displays offer sent to MediaSense having only audio m-lines, when the media-type audio command is configured.

        
        Sent:
        INVITE sip:54321@9.45.38.39:36212 SIP/2.0
        Via: SIP/2.0/UDP 9.41.36.15:5060;branch=z9hG4bK2216B
        X-Cisco-Recording-Participant: sip:4321@9.45.38.39;media-index="0"
        X-Cisco-Recording-Participant: sip:1111000010@9.45.38.39;media-index="1"
        From: <sip:9.41.36.15>;tag=A2C74-5D9
        To: <sip:54321@9.45.38.39>……
        Content-Type: application/sdp
        Content-Disposition: session;handling=required
        Content-Length: 337
        
        v=0
        o=CiscoSystemsSIP-GW-UserAgent 9849 5909 IN IP4 9.41.36.15
        s=SIP Call
        c=IN IP4 9.41.36.15
        t=0 0
        m=audio 16392 RTP/AVP 0 19
        c=IN IP4 9.41.36.15
        a=rtpmap:0 PCMU/8000
        a=rtpmap:19 CN/8000
        a=ptime:20
        a=sendonly
        m=audio 16394 RTP/AVP 0 19
        c=IN IP4 9.41.36.15
        a=rtpmap:0 PCMU/8000
        a=rtpmap:19 CN/8000
        a=ptime:20
        a=sendonly
        

        Response from CUBE has inactive video m-lines.

        Received:  
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP 9.41.36.15:5060;branch=z9hG4bK2216B
        …..
        v=0
        …
        m=audio 36600 RTP/AVP 0
        c=IN IP4 9.45.38.39
        a=rtpmap:0 PCMU/8000
        a=ptime:20
        a=recvonly
        m=audio 36602 RTP/AVP 0
        c=IN IP4 9.45.38.39
        a=rtpmap:0 PCMU/8000
        a=ptime:20
        a=recvonly
        m=video 0 RTP/AVP 98
        c=IN IP4 9.45.38.39
        b=TIAS:1500000
        a=rtpmap:98 H264/90000
        a=fmtp:98 profile-level-id=420015
        a=inactive
        m=video 0 RTP/AVP 98
        c=IN IP4 9.45.38.39
        b=TIAS:1500000
        a=rtpmap:98 H264/90000
        a=fmtp:98 profile-level-id=420015
        a=inactive
        
        Step 13   Enter one of the following:
        • debug ccsip all
        • debug voip recmsp all
        • debug voip ccapi all
        • debug voip fpi all (for ASR devices only)

        Displays detailed debug messages.

        For Audio:

        Media forking initialized:

        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_trigger_media_forking: MF: Recv Ack..
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_trigger_media_forking: MF: Recv Ack & it's Anchor leg. Start MF.
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_ipip_media_forking_preprocess_event: MF: initial-call. State = 1 & posting the event E_IPIP_MEDIA_FORKING_CALLSETUP_IND
        

        Media forking started:

        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_ipip_media_service_get_event_data: Event id = 30
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Function/sipSPIUisValidCcb: 
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Function/ccsip_is_valid_ccb: 
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_ipip_media_forking: MF: Current State = 1, event =30
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_ipip_media_forking: MF: State & Event combination is cracked..
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Function/sipSPIGetMainStream: 
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Function/sipSPIGetMainStream: 
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_ipip_media_forking_precondition: MF: Can be started with current config.
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_ipip_media_forking_BuildMediaRecParticipant: MF: Populate rec parti header from this leg.
        

        Forking header populated:

        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_get_recording_participant_header: MF: X-Cisco header is RPID..
        

        Media forking setup record session is successful:

        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_get_recording_participant_header: MF: Building SIP URL..
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_get_recording_participant_header: MF: Sipuser = 98459845
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_get_recording_participant_header: MF: Host = 9.42.30.34
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Function/sipSPIGetFirstStream: 
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Function/voip_media_dir_to_cc_media_dir: 
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_ipip_media_forking_BuildMediaRecStream: MF: direction type =3 3
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_ipip_media_forking_BuildMediaRecStream: MF: callid 103 set to nearend..
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_ipip_media_forking_BuildMediaRecStream: MF: dtmf is inband
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_ipip_media_forking_BuildMediaRecStream: MF: First element..
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_ipip_media_forking_BuildMediaRecParticipant: MF: First element..
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_ipip_media_forking_BuildMediaRecParticipant: MF: Populate rec parti header from peer leg.
        *Jun 15 10:37:55.404: //104/3E7E90AE8006/SIP/Info/ccsip_get_recording_participant_header: MF: X-Cisco header is RPID..
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_ipip_media_forking_write_to_TDContainer: MF: Data written to TD Container..
        *Jun 15 10:37:55.404: //-1/xxxxxxxxxxxx/Event/recmsp_api_setup_session: Event: E_REC_SETUP_REQ anchor call ID:103, msp call ID:105 infunction recmsp_api_setup_session
        *Jun 15 10:37:55.404: //-1/xxxxxxxxxxxx/Inout/recmsp_api_setup_session:  Exit with Success
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/act_sip_mf_idle_callsetup_ind: MF: setup_record_session is success..
        

        Media forking forked stream started:

        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/sipSPIMFChangeState: MF: Prev state = 1 & New state = 2
        *Jun 15 10:37:55.404: //103/3E7E90AE8006/SIP/Info/ccsip_gen_service_process_event: MF: 30 event handled.
        *Jun 15 10:37:55.406: //106/000000000000/SIP/Info/ccsip_call_setup_request: Set Protocol information
        *Jun 15 10:37:55.406: //106/xxxxxxxxxxxx/CCAPI/cc_set_post_tagdata:
        *Jun 15 10:37:55.406: //106/000000000000/SIP/Info/ccsip_ipip_media_forking_read_from_TDContainer: MF: Data read from TD container..
        *Jun 15 10:37:55.406: //106/000000000000/SIP/Info/ccsip_ipip_media_forking_forked_leg_config: MF: MSP callid = 105
        *Jun 15 10:37:55.406: //106/000000000000/SIP/Info/ccsip_ipip_media_forking_forked_leg_config: MF: Overwriting the GUID with the value got from MSP.
        *Jun 15 10:37:55.406: //106/000000000000/SIP/Info/ccsip_iwf_handle_peer_event: 
        *Jun 15 10:37:55.406: //106/000000000000/SIP/Info/ccsip_iwf_map_ccapi_event_to_iwf_event: Event Category: 1, Event Id: 179
        *Jun 15 10:37:55.406: //106/000000000000/SIP/Info/ccsip_iwf_process_event:  
        *Jun 15 10:37:55.406: //106/000000000000/SIP/Function/sipSPIUisValidCcb: 
        *Jun 15 10:37:55.406: //106/3E7E90AE8006/SIP/Info/ccsip_ipip_media_forking_add_forking_stream: MF: Forked stream added..
        *Jun 15 10:37:55.406: //106/3E7E90AE8006/SIP/Info/ccsip_ipip_media_forking_read_from_TDContainer: MF: Data read from TD container..
        *Jun 15 10:37:55.406: //106/3E7E90AE8006/SIP/Function/sipSPIGetFirstStream: 
        *Jun 15 10:37:55.406: //106/3E7E90AE8006/SIP/Info/ccsip_ipip_media_forking_Display_TDContainerData: ** DISPLAY REC PART ***
        *Jun 15 10:37:55.406: //106/3E7E90AE8006/SIP/Info/ccsip_ipip_media_forking_Display_TDContainerData: recorder tag = 5
        
        For Video:

        Media Forking Initialized:

        *Mar 19 16:40:01.784 IST: //522/34BF0A000000/SIP/Info/notify/32768/ccsip_trigger_media_forking: MF: Recv Ack & it's Anchor leg. Start MF.
        *Mar 19 16:40:01.784 IST: //522/34BF0A000000/SIP/Info/info/32768/ccsip_ipip_media_forking_preprocess_event: MF: initial-call. State = 1 & posting the event E_IPIP_MEDIA_FORKING_CALLSETUP_IND
        

        Media forking started:

         *Mar 19 16:40:01.784 IST: //522/34BF0A000000/SIP/Info/info/36864/ccsip_ipip_media_forking: MF: Current State = 1, event =31
        *Mar 19 16:40:01.784 IST: //522/34BF0A000000/SIP/Info/info/36864/ccsip_ipip_media_forking: MF: State & Event combination is cracked..
        *Mar 19 16:40:01.784 IST: //522/34BF0A000000/SIP/Function/sipSPIGetMainStream:
         *Mar 19 16:40:01.784 IST: //522/34BF0A000000/SIP/Function/sipSPIGetMainStream: 
        *Mar 19 16:40:01.787 IST: //522/34BF0A000000/SIP/Info/info/34816/ccsip_ipip_media_forking_precondition: MF: Can be started with current config.
        *Mar 19 16:40:01.787 IST: //-1/xxxxxxxxxxxx/Event/recmsp_api_create_session: Event: E_REC_CREATE_SESSION anchor call ID:522, msp call ID:526
        *Mar 19 16:40:01.787 IST: //-1/xxxxxxxxxxxx/Inout/recmsp_api_create_session:  Exit with Success
        

        Recording participant for anchor leg:

        //522/34BF0A000000/SIP/Info/verbose/32768/ccsip_ipip_media_forking_BuildMediaRecParticipant: MF: Populate rec parti header from this leg.
        *Mar 19 16:40:01.788 IST: //522/34BF0A000000/SIP/Info/info/33792/ccsip_get_recording_participant_header: MF: X-Cisco header is PAI..
        

        Adding an audio stream:

        *Mar 19 16:40:01.788 IST: //522/34BF0A000000/SIP/Function/sipSPIGetFirstStream: 
        *Mar 19 16:40:01.788 IST: //522/34BF0A000000/SIP/Info/verbose/32768/ccsip_ipip_media_forking_BuildMediaRecStream: MF: Adding a Audio stream..
        *Mar 19 16:40:01.789 IST: //522/34BF0A000000/SIP/Function/voip_media_dir_to_cc_media_dir: 
        *Mar 19 16:40:01.789 IST: //522/34BF0A000000/SIP/Info/info/32768/ccsip_ipip_media_forking_BuildAudioRecStream: MF: direction type =3 3
        *Mar 19 16:40:01.789 IST: //522/34BF0A000000/SIP/Info/info/32768/ccsip_ipip_media_forking_BuildAudioRecStream: MF: callid 522 set to nearend..
        *Mar 19 16:40:01.789 IST: //522/34BF0A000000/SIP/Info/info/32768/ccsip_ipip_media_forking_BuildAudioRecStream: MF: This rcstream has 522 callid 
        *Mar 19 16:40:01.789 IST: //522/34BF0A000000/SIP/Info/verbose/32768/ccsip_ipip_media_forking_BuildAudioRecStream: MF: Setting data for audio stream..
        *Mar 19 16:40:01.789 IST: //522/34BF0A000000/SIP/Info/info/32800/ccsip_ipip_media_forking_BuildAudioRecStream: MF: dtmf is inband
        .
        

        Video forking:

        *Mar 19 16:40:01.789 IST: //522/34BF0A000000/SIP/Function/sipSPIGetVideoStream: 
        *Mar 19 16:40:01.789 IST: //522/34BF0A000000/SIP/Info/verbose/32772/ccsip_ipip_media_forking_BuildMediaRecStream: MF: video_codec present,Continue with Video Forking..
        
        For Video

        Additional References for Network-Based Recording

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        MediaSense Installation and Administration Guide

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