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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 recording session (RS) is established over SIP from the SRC to the SRS.

The 

Table 1: Supported RFCs

In the following figure, a basic call is established between SIP phone 1 and SIP phone 2 through the

Figure 1: 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 

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  to record the call.
• Initiate recording of a specific call using CLI/EMA and specifying the appropriate GCID.

PSX Configuration

The recording-criteria determine which sessions to record and target 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).
  • Transport
  • IP V4/V6 address and port
• Contains
• SRS Groups—contains multiple Recording profiles for SRS redundancy (up to 8).
  • Transport
  • IP V4/V6 address port
  • Encryption data (for SRTP)  
  • IP TG to be used by the SBC for RS session
• Contains data of multiple SRS servers
• 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

You can initiated a recording session for a specific call by providing its GCID through a CLI "request" command. In the command you provide other details for the recording such as the IP address/port or FQDN/port for the corresponding session recorder. Refer to Request Global - CLI for more information. 

To enable/disable the SIPREC feature on a zone, use following syntax:

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

Refer to Zone - SIP Sig Port - CLI for more information.

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

% request global siprec startRecord gcid <GCID> callLeg ingress numOfStreams <Number of recorders  1 or 2> srsIpAddress <SRS IP ADDRESS> srsFqdn1 <FQDN> srsPort <SRS PORT> transport <tcp | udp> trunkGroup <TRUNK GROUP NAME> srsIpAddress2 <SRS IP ADDRESS> srsFqdn2 <secondary FQDN> srsPort2 <SRS Port> transport2 <tcp | udp> trunkGroup2 <SIP Trunk Group> 

% request global siprec stopRecord gcid <GCID> recorderAddress <IP Address> recorderFqdn <FQDN> recorderPort <Port Number> recorderId <recording session ID>

Through CLI you can also list information on the calls currently being recorded such as the GCID, RTP addresses and the SRS IP address or FQDN.  To view SIPREC status, use  the following CLI syntax:

> show table global SipRecStatus                
       RECORDER          RECORDING  RX RTP           TX RTP           RECORDER  RECORDER
GCID   ADDRESS           LEG        ADDRESS          ADDRESS          FQDN      ID
555281 10.11.12.13:5060  ingress   10.11.12.13:8000  10.11.12.13:8002           1614567899

Recording Procedure

Once the 

• Includes a “+sip.src” feature tag extension in the Contact URI
• Includes an Options tag “siprec” in INVITE towards SRS.
• Includes 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 attributes of the communication session (CS).
• <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.

Supported SIPREC Features

The 

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

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  supports recording SIP and SIP-I calls.
• If the 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 pings. 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 

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 two recordings on egress call-leg.

Figure 2: An Example of Simultaneous Session Recording

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 SRS

The 

With this feature, the

• 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 configuration in the srsGroupData.
• Supports sending SRTP packets (encrypted media streams) as received from the endpoints (relay/pass-through).
• Supports sending SRTP packets with a different encryption technique using the crypto suite profile configured in the srsGroupData.
• Provides support to terminate an 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 crypto suite profile negotiated between the SBC and the UE.
  • The decrypted media packets towards the SRS can be:
    • Encrypted using the same negotiated crypto suite profile.
    • Encrypted using different crypto suite profile as provided by the srsGroupData.
    • Send only decrypted media packets (RTP) towards the SRS based on the configuration (CS crypto is ignored).

The following two options in srsGroupData control SRTP support:

• srtp: Specifies whether SRTP is enabled for the SRS or not.
• cryptoSuiteProfile: If SRTP is enabled, the SBC uses the encryption technique called for in the specified profile for the recording session towards the SRS.

When SRTP is disabled for the SRS, the 

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

• If a cryptoSuiteProfile is configured for the SRS, the

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

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   sends RTP packets. 

When the CS is using SRTP pass-through:

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

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

When the CS is an SRTP terminated call:

• If a cryptoSuiteProfile is configured for the SRS, the SBC sends the SRTP packets using the cryptoSuiteProfile on the recorded leg towards the SRS. 
  
• If a 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 provides the capability to configure the headers that are mapped from the target call leg to 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. The "To" header and "to-tag" are also copied from local information (as the to-tag is not present in the INVITE).
• In case of a SIPREC trigger during a 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 a CLI-triggered recording, the existing implementation of sending predefined information in metadata XML remains the 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 names.

Figure 3: Call Flow

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SIPREC on SBC Cloud-Native Edition (CNe)

The SBC CNe is enhanced to support SIP Recording (SIPREC) when deployed in a Cloud-Native Network Function (CNF) environment. SIPREC sends media and signaling info to the SIP Recording Servers, which allows the user to monitor quality of service, store call information for quality analysis, and comply with regulations. The SBC CNe supports SIPREC in both Packet Front End (PFE) and non-PFE deployments.

This feature adds support for the Global Unique Identifier (GUID) alongside the existing Global Call Identifier (GCID) on the SBC CNe. GUID is superior to the existing GCID because GUID acts as a global identifier for a CNF cluster for calls, whereas GCID can only uniquely identify calls within a single Session Control (SC) pod. For example, different calls can share a common GCID across multiple SC pods, or multiple SC pods can share the same GCID while hosting different calls.

Additionally, when triggering SIPREC status and action commands, use GUID to identify all pods in a CNF cluster by setting podName to <ALL>; meanwhile, GCID requires that you identify each individual SC pod when triggering SIPREC status and action commands. Therefore, Ribbon recommends using the GUID on the SBC CNe.

SIPREC status and action commands are updated to accept GUID as the key on the SBC CNe. As well, statistics commands are updated to support GUID.

For more information about SIPREC on SBC CNe, refer to:

• Request Global - CLI
• Global - SIP Rec Status
• Global - SIPrec
• Global - Media Capture

Known Issues

This section lists all known issues related to SIPREC on the

SRS Group roundRobin loadDistribution Issue

Problem Statement: In the SRS Group, the "roundRobin" loadDistribution scheme does not function properly when the following two criteria are both met:

• A call that matches the SIPREC recording criteria has multiple routes;
• The number of SRS IPs is exactly divisible by the number of routes.

Under standard operations when "roundrobin" is selected as the loadDistribution scheme for the SRS Group, the SRS IPs are shifted in the srsGroupProfile each time the SIPREC criteria is matched.

However, when the number of SRS IPs is exactly divisible by the number of routes, the SRS IPs jump by the number of routes. This results in SIPREC skipping certain SRS IPs.

Example: If the srsGroupProfile is configured with six SRS Servers (SRS_IP1, SRS_IP2, SRS_IP3, SRS_IP4, SRS_IP5, SRS_IP6) and two routes (Route#0, Route#1), then SIPREC will behave as follows when the recording criteria is matched at ingress TG:

1. For Call#1, Route#0, 0th entry will be SRS_IP1
2. For Call#1, Route#1, 0th entry will be SRS_IP2
3. For Call#2, Route#0, 0th entry will be SRS_IP3
4. For Call#2, Route#1, 0th entry will be SRS_IP4
5. For Call#3, Route#0, 0th entry will be SRS_IP5
6. For Call#3, Route#1, 0th entry will be SRS_IP6
7. For Call#4, Route#0, 0th entry will be SRS_IP1
8. For Call#4, Route#1, 0th entry will be SRS_IP2

Assuming that Route#0 is always successful, then only SRS_IP1, SRS_IP3, and SRS_IP5 will record. Therefore, SRS_IP2, SRS_IP4, and SRS_IP6 will always get skipped because the number of SRS IPs (six) is exactly divisible by the number of routes (two).

Solution:Configure the number of SRS IPs so that they are not exactly divisible by the number of routes returned for the call. Users can accomplish this by adding a mock IP or a repeated SRS IP to the srsGroupProfile. Additionally, users are recommended to update the number of SRS IPs to a prime number (ex: 5, 7, 11, 13, etc.) to minimize the chance that the number of SRS IPs is divisible by the number of routes.

Returning to the example above, the issue is resolved if a seventh SRS IP is added (SRS_IP7). Since the number of SRS IPs is no longer exactly divisible by the number of routes, the recordings flow properly, as follows: SRS_IP1, SRS_IP3, SRS_IP5, SRS_IP7, SRS_IP2, SRS_IP4, SRS_IP6, SRS_IP1, and so on.

Example SIP INVITE

An example 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>