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Overview

The SIPREC protocol defines the interaction between a Session Recording Client (SRC) and a Session Recording Server (SRS), and controls the recording of media transmitted in the context of a communications session (CS) between multiple user agents. The Recoding Session (RS) is established over SIP from SRC to SRS.

The SBC  

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supports SIPREC towards multiple recorders based on the Internet Engineering Task Force (IETF) standard. The SBC acts as an SRC sending recording sessions to a third-party SRS when a configured recording-criteria is met.

Multiexcerpt include
MultiExcerptName Recording Limit PageWithExcerpt Session Recording Support

Caption
0 Table 1 Supported RFCs
Category RFCs Use Cases and Requirements for SIP-Based Media Recording (SIPREC) RFC 6341 Media Recording Architecture RFC 7245  Session Initiation Protocol (SIP) Recording Metadata RFC R0FC 7865 Session Recording Protocol RFC 7866 Session Initiation Protocol (SIP) Recording Call Flows RFC 8068

In the figure SBC SIP Recording Strategy, the basic call is established between SIP phone 1 and SIP phone 2 through the SBC

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, which is known as communication session (CS). The SBC establishes The 

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establishes an RS based on CS towards SRS. The SBC and SRS may exist in the same or different administrative domains. 

Caption
0 Figure 1 SBC SIP Recording Strategy 3 SBC SIP Recording Strategy

Configuring SIPREC Based Recording

The two methods to trigger a call recording are:

• A call matches call recording criteria causing the Policy Server to trigger the SBC the 

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  to record the call.
• Initiate recording via CLI using GCID.

PSX Configuration

The recording-criteria determine, which sessions to record, SRS information along with other operational options.

The PSX/ERE uses the following configurable objects when determining whether a call needs to be recorded or not:

• Recording Criteria—contain the rules to match for invoking call recording (this is same for SIPREC and MCT).
• Recorder Profile—contains information about the recorder to use for a particular Recording Entity (obsolete—used in older versions when SRS redundancy is not supported).
• Contains
• Transport
• IP V4/V6 address and port
• SRS Groups—contains multiple Recoding profiles for SRS redundancy (up to 8).
• Contains data of multiple SRS servers
• Transport
• IP V4/V6 address port
• Encryption data (for SRTP)  
• IP TG to be used by the SBC for RS session
• Recording Cluster profile—contains multiple SRS Groups for simultaneous recording (up to 4).

Recording Criteria

The need to record a call is decided from the PSX based on the following criteria in the given order of priority:

• Recorder type
• SIPREC
• MCT
• Next hop signaling IP address
• Previous hop signaling IP address
• Calling Party Number
• Called Party number
• Ingress TG ID
• Egress TG ID
• GATEWAY

For detailed configuration information, refer to the section Deploying SBC For SIPREC

CLI Triggers

• Recording RTP session can be initiated for a call by providing its GCID through CLI. The user provides the IP Address/port for the corresponding session recorder through the same CLI command.
• Listing the calls currently being recorded using a CLI command. The SBC displays The 

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  displays their GCID, and the RTP destinations and the SRS IP address.

Once SBC determines that the call must be recorded, it initiates the SIP INVITE towards the SRS specified in the recording criteria.

• Includes a “+sip.src” feature tag extension in the Contact URI
• Options Tag “siprec” in INVITE towards SRS.
• Two m= lines one for the Rx stream and one for the Tx stream.
• The attribute “a=label” identifies the streams in the metadata.
• Adds the RS-call specific data in the rs-metadata XML body
• This metadata provides information on the participants of the communication session
• <gcid> the Global call ID used in the SBC
• Other SIP headers with their corresponding XML tags are added as per configuration
• Sends the SDP offer with the same Codec of the call leg that is being recorded towards the SRS.
• When the SRS replies with its media IP/port information in a 200 OK, the SBC transparently duplicates the packets coming/going to the UE towards the SRS using the same Codec as the Original stream.

For configuring SIPREC feature, refer to the section Deploying SBC For SIPREC.

Supported Features

The SBC supports following SIPREC features:

• The SIPREC is supported on UDP, TCP, and TLS protocols.
• The SBC supports recording SIP and SIP-I calls.
• If Options PING mechanism is configured on the IP peer, it is used as a keep-alive mechanism for all the SRSs.
• Troubleshooting a SIPREC recorded call using MCT is only possible when initiated through CLI.
• SRS Redundancy
• Simultaneous recording
• Dynamically programmable metadata content
• The SBC records the SIPREC recording information in the CDR.
  

To enable/disable SIPREC feature, use following syntax:

% set addressContext <ADDRESS-CONTEXT> zone <ZONE> sipSigPort <SIP SIGNALLING PORT> siprec <disabled|enabled>

To start/stop a recording, the following CLI syntax applies:

% request global sipRec startRecord gcid <GCID> callLeg ingress numOfStreams <Number of recorders  1 or 2> srsIpAddress <SRS IP ADDRESS> srsPort <SRS PORT> transport <tcp | udp> trunkGroup <TRUNK GROUP NAME> srsIpAddress2 <SRS IP ADDRESS> srsPort2 <SRS Port> transport2 <tcp | udp> trunkGroup2 <SIP Trunk Group> 
% request global sipRec stopRecord gcid <GCID> recorderAddress <IP Address> recorderPort <Port Number>

To view SIPREC status, use CLI syntax:

> show table global SipRecStatus
       RECORDER           RX RTP            TX RTP            RECORDING  
GCID   ADDRESS            ADDRESS           ADDRESS           LEG
1      10.11.12.13:5060   10.11.12.13:8000  10.11.12.13:8002  ingress

Refer to Zone - SIP Sig Port - CLI and Request Global - CLI pages for CLI command details.

SRS Redundancy

The SBC supports the concept of Primary and secondary SRS servers for redundancy. It supports multiple (up to 8) SRS servers in a SRS Group. All of them can be active at any point of time.

To enable/disable SIPREC feature, use following syntax:

% set addressContext <ADDRESS-CONTEXT> zone <ZONE> sipSigPort <SIP SIGNALLING PORT> siprec <disabled|enabled>

To start/stop a recording, the following CLI syntax applies:

% request global sipRec startRecord gcid <GCID> callLeg ingress numOfStreams <Number of recorders  1 or 2> srsIpAddress <SRS IP ADDRESS> srsPort <SRS PORT> transport <tcp | udp> trunkGroup <TRUNK GROUP NAME> srsIpAddress2 <SRS IP ADDRESS> srsPort2 <SRS Port> transport2 <tcp | udp> trunkGroup2 <SIP Trunk Group> 
% request global sipRec stopRecord gcid <GCID> recorderAddress <IP Address> recorderPort <Port Number>

To view SIPREC status, use CLI syntax:

> show table global SipRecStatus
       RECORDER           RX RTP            TX RTP            RECORDING  
GCID   ADDRESS            ADDRESS           ADDRESS           LEG
1      10.11.12.13:5060   10.11.12.13:8000  10.11.12.13:8002  ingress

Refer to Zone - SIP Sig Port - CLI and Request Global - CLI pages for CLI command details.

Recording Procedure

Once the 

• Includes a “+sip.src” feature tag extension in the Contact URI
• Options Tag “siprec” in INVITE towards SRS.
• Two m= lines one for the Rx stream and one for the Tx stream.
• The attribute “a=label” identifies the streams in the metadata.
• Adds the RS-call specific data in the rs-metadata XML body
• This metadata provides information on the participants of the communication session
• <gcid> the Global call ID used in the SBC
• Other SIP headers with their corresponding XML tags are added as per configuration
• Sends the SDP offer with the same Codec of the call leg that is being recorded towards the SRS.
• When the SRS replies with its media IP/port information in a 200 OK, the SBC transparently duplicates the packets coming/going to the UE towards the SRS using the same Codec as the Original stream.

For configuring SIPREC feature, refer to the section Deploying SBC For SIPREC.

Supported Features

The 

• The SIPREC is supported on UDP, TCP, and TLS protocols.
• The 

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  supports recording SIP and SIP-I calls.
• If Options PING mechanism is configured on the IP peer, it is used as a keep-alive mechanism for all the SRSs.
• Troubleshooting a SIPREC recorded call using MCT is only possible when initiated through CLI.
• SRS Redundancy
• Simultaneous recording
• Dynamically programmable metadata content
• The SBC records the SIPREC recording information in the CDR.
  

SRS Redundancy

The 

• The 

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  can load balance among the recorders in the SRS Group in sequential or round-robin fashion.
• The 

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  also supports crank-back mechanism where it tries the next SRS if an SRS does not respond to the RS INVITE or rejects the RS INVITE with a 4xx.
• The 

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  also maintains the health check of the SRS servers using Options PING. If an SRS is marked not reachable, the SBC automatically uses the next server in the SRS group for the next RS session.

Simultaneous Session Recording - SIP Ingress and Egress Legs

The SBC The 

The 

Example of Simultaneous Session Recording

Below is a diagram illustrating the use case of a simultaneous SIPREC, with first two recordings on Ingres call-leg, and the next 2 two recordings on egress call-leg.

For more information on parameter configurations and CDR field descriptions refer to:

CLI and EMA:

• Request Global - CLI
• Servers - SRS Group Profile - CLI
• Servers - Call Recording Criteria - CLI
• Servers - SRS Group Profile - CLI
• Zone - SIP Sig Port - CLI
• Global - SIPrec
• Global - Servers - Srs Group Profile

Alarms:

• sonusSbxSipRecSrsSelectionFailedNotification - MINOR
• sonusSbxSIPRecLicenseNotAvailable - MINOR

CDR:

• CDR Field Descriptions

SRTP Support for SIPREC Towards SRSSRS

The 

• The 

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  The SBC sends the SRTP streams as received from the endpoints.
• The SBC The 

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  is configured to perform a different encryption (using dedicated crypto suite profile) towards the Session Recording Server (SRS).
• The 

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  The SBC is configured to send plain RTP packets even when the original Communication Session (CS) is an SRTP call.

With this feature, the SBC

• Supports SRTP (encrypted streams) towards SIPREC server.
• Provides SIPREC functionality with SRTP calls.
• Provides flexibility by supporting both SRTP or RTP towards the SRS based on the input control in the srsGroupData.
• Supports sending SRTP packets (encrypted media streams) as received from the endpoints (relay/pass-through).
• Supports sending SRTP packets with different encryption techniques using new crypto suite profile configured in the srsGroupData.
• Provides support to terminate original SRTP call at the SBC for the SIPREC, such that:
  • The encrypted media packets are decrypted at the SBC and are re-encrypted again towards the user endpoint using the negotiated crypto suite profile (CS) between the SBC and the UE.
  • The decrypted media packets towards SRS can be:
    • Encrypted using the same CS negotiated crypto suite profile.
    • Encrypted using different crypto suite profile as provided by the srsGroupData.
    • Send decrypted media packets (RTP) only towards SRS based on the configuration (CS crypto is ignored).

The following two options are added to the srsGroupData:

• srtp: Specifies whether SRTP is enabled for the SRS or not.
• cryptoSuiteProfile: If SRTP is enabled, encrypt recording session using this crypto details.

When SRTP is disabled for the SRS, the SBC the 

When SRTP is enabled for the SRS and if the CS leg that is recorded is not using SRTP:

• If cryptoSuiteProfile is configured for the SRS, the S

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   sends BC sends the SRTP packets using the cryptoSuiteProfile on the recorded leg towards the SRS. 
   
• If cryptoSuiteProfile is not configured for the SRS, the

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   sends SBC sends the RTP packets. 

When the CS is using SRTP pass-through:

• If the call is recorded and SRTP is enabled for the SRS, the SBC the 

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  relays the SRTP packets as recevied from the user endpoint.
• If SRTP is enabled for the SRS and cryptoSuiteProfile is configured, the SBC re-encrypts the media using the configured cryptoSuiteProfile.

When the CS is an SRTP terminated call:

• If cryptoSuiteProfile is configured for the SRS, the SBC sends the SRTP packets using the cryptoSuiteProfile on the recorded leg towards the SRS. 
  
• If cryptoSuiteProfile is not configured for the SRS, the SBC uses the cryptoSuiteProfile from the CS and sends SRTP packets.

Dynamically Programmable Metadata Content

The 

• The profile sipRecMetaDataProfile is introduced to the services to provide the capability to configure the headers that are mapped from the target call leg to the XML and the corresponding metadata XML element name.
• In case of a basic call, all information is copied from the initial-INVITE message on the leg where the tap is, to the metadata XML. However, "To" header and "to-tag" is copied additionally from the local information (as to-tag does not present in the INVITE).
• In case of SIPREC trigger during REFER based transfer, irrespective of where the SIPREC tap is, all information is copied from the initial-INVITE of the new call leg towards the transfer target (C party).
• In case of CLI triggered recording, the existing implementation of sending predefined information in metadata XML remains same (gcid, call-id, from, to). The new configuration of header-metadata mapping is not considered in this scenario.

The following call flow diagram displays the XML tag name.

Example SIP INVITE

The following is an An example of SIP INVITE is shown below:

INVITE sip:SIPREC-SRS@10.54.80.8:51802 SIP/2.0
Via: SIP/2.0/UDP 10.34.171.39:5060;branch=z9hG4bK00B00021f4cdc2590c2
From: "SIPREC-SRC" <sip:SIPREC-SRC@10.34.171.39>;tag=gK00000237
To: "SIPREC-SRS" <sip:SIPREC-SRS@10.54.80.8>
Call-ID: 35651585_16777218_133945398@10.34.171.39
CSeq: 787532 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/rs-metadata-request,application/rs-metadata
Contact: "SIPREC-SRC" <sip:SIPREC-SRC@10.34.171.39:5060>;+sip.src
Require: siprec
Supported: timer,100rel
Session-Expires: 1800
Min-SE: 90
Content-Length:  4560
Content-Type: multipart/mixed;boundary=sonus-content-delim
MIME-Version: 1.0

--sonus-content-delim
Content-Disposition: session; handling=required
Content-Length:   296
Content-Type: application/sdp

v=0
o=Sonus_UAC 748003 60371 IN IP4 10.34.171.39
s=SIP Media Capabilities
t=0 0
m=audio 

1052 RTP/

SAVP 0
c=IN IP4 10.

54.

4.

51
a=label:1
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:k3NkQB3Tkr23twOMMjd8YjvLI/XPdgE+a1D8FDho
a=rtpmap:0 PCMU/8000
a=sendonly
a=maxptime:10
m=audio 

1050 RTP/

SAVP 0
c=IN IP4 10.

54.

4.

51
a=label:2
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:k3NkQB3Tkr23twOMMjd8YjvLI/XPdgE+a1D8FDho
a=rtpmap:0 PCMU/8000
a=sendonly
a=maxptime:10

--sonus-content-delim
Content-Disposition: recording-session
Content-Length:  3976
Content-Type: application/rs-metadata+xml

<?xml version="1.0" encoding="UTF-8"?>
<recording xmlns='urn:ietf:params:xml:ns:recording:1'>
    <datamode>complete</datamode>
    <group group_id="OTIxYzk4MDAtN2RkYy0xMA==">
        <associate-time>2018-08-09T08:31:44Z</associate-time>
        <callData xmlns='http://ribboncommunications.com/siprec/calldata'>
            <xTo>&lt;sip:+1999@10.54.80.8:51801;user=phone&gt;;tag=1</xTo>
            <xVia>SIP/2.0/UDP 10.34.171.34:5060;branch=z9hG4bK00B0000a25afeb7eee5</xVia>
            <xCSeq>844797 INVITE</xCSeq>
            <xFrom>"sipp" <sip:sanrayana@10.34.171.34>;tag=gK0000011e</xFrom>
            <xContentType>application/sdp</xContentType>
            <xMaxForwards>70</xMaxForwards>
            <srsgrpId>GR1</srsgrpId>
            <xAcceptContact>*;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"</xAcceptContact>
            <xPPreferredIdentity>"sipp" <sip:sanrayana@10.54.80.8:5061></xPPreferredIdentity>
            <mprofileVers>v1.0</mprofileVers>
            <gcid>35651585</gcid>
        </callData>
    </group>
    <session session_id="OTIxYzlhM2UtN2RkYy0xMA==">
        <group-ref>OTIxYzk4MDAtN2RkYy0xMA==</group-ref>
        <start-time>2018-08-09T08:31:44Z</start-time>
    </session>
    <participant participant_id="OTIxYzk4MDEtN2RkYy0xMA==">
        <nameID aor="sanrayana@10.34.171.34:5060">
            <name xml:lang="en">sipp</name>
        </nameID>
    </participant>
    <participant participant_id="OTIxYzk4MDItN2RkYy0xMA==">
        <nameID aor="+1999@10.54.80.8">
            <name xml:lang="en"> </name>
        </nameID>
    </participant>
    <stream stream_id="OTIxYzk4MDQtN2RkYy0xMA==" session_id="OTIxYzlhM2UtN2RkYy0xMA==">
        <label>1</label>
        <associate-time>2018-08-09T08:31:44Z</associate-time>
    </stream>
    <stream stream_id="OTIxYzk4MDUtN2RkYy0xMA==" session_id="OTIxYzlhM2UtN2RkYy0xMA==">
        <label>2</label>
        <associate-time>2018-08-09T08:31:44Z</associate-time>
    </stream>
    <sessionrecordingassoc session_id="OTIxYzlhM2UtN2RkYy0xMA==">
        <associate-time>2018-08-09T08:31:44Z</associate-time>
    </sessionrecordingassoc>
    <participantsessionassoc participant_id="OTIxYzk4MDEtN2RkYy0xMA==" session_id="OTIxYzlhM2UtN2RkYy0xMA==">
        <associate-time>2018-08-09T08:31:44Z</associate-time>
    </participantsessionassoc>
    <participantsessionassoc participant_id="OTIxYzk4MDItN2RkYy0xMA==" session_id="OTIxYzlhM2UtN2RkYy0xMA==">
        <associate-time>2018-08-09T08:31:44Z</associate-time>
    </participantsessionassoc>
    <participantstreamassoc participant_id="OTIxYzk4MDEtN2RkYy0xMA==">
        <send>OTIxYzk4MDQtN2RkYy0xMA==</send>
        <recv>OTIxYzk4MDUtN2RkYy0xMA==</recv>
    </participantstreamassoc>
    <participantstreamassoc participant_id="OTIxYzk4MDItN2RkYy0xMA==">
        <send>OTIxYzk4MDUtN2RkYy0xMA==</send>
        <recv>OTIxYzk4MDQtN2RkYy0xMA==</recv>
    </participantstreamassoc>
</recording>