RFC 7866
Session Recording Protocol
Session Recording Protocol 🎀
Okay bestie, let's start with the header, because it's giving main character energy 👑. RFC 7866 is a Standards Track document from May 2016, and it was published by the IETF, with ISSN 2070-1721. Five authors worked on it: L. Portman from NICE Systems, H. Lum (the editor, from Genesys), C. Eckel from Cisco, A. Johnston from the Illinois Institute of Technology, and A. Hutton from Unify 💖. The header doesn't list any RFCs that it obsoletes or updates, so there's nothing to track there 🦋.
Abstract ✨
So this doc explains how to use SIP, SDP, and RTP to send real-time media and metadata from a Communication Session (CS) over to a recording device 🎧. The Session Recording Protocol sets up a Recording Session (RS) between a Session Recording Client (SRC), which sits on the path of the CS, and a Session Recording Server (SRS) at the recording device 🫶. It only covers active recording, where the SRC deliberately streams media to an SRS and every participating user agent (UA) gets told that recording is happening 🔥. Passive recording is a different vibe. That's where a device grabs media straight off the network, like with port-mirroring, and it's outside the scope of this doc 🙈. Lawful intercept is also outside the scope 💀.
Status of This Memo ✨
This is an Internet Standards Track document 🌈.
It's a product of the Internet Engineering Task Force (IETF), and it reflects consensus from the IETF community 🫶. It got public review and was approved for publication by the Internet Engineering Steering Group (IESG). For more on Internet Standards, check Section 2 of RFC 7841 💜.
For the current status, any errata, and how to send feedback, go to http://www.rfc-editor.org/info/rfc7866 🍓.
Copyright Notice 💅
Copyright (c) 2016 IETF Trust and the persons identified as the document authors. All rights reserved ✨.
This doc is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents, which you can find at http://trustee.ietf.org/license-info as they stood on the publication date 📜. Please read those carefully, because they spell out your rights and restrictions for this document 🙏. Code Components pulled out of this document must include Simplified BSD License text, as described in Section 4.e of the Trust Legal Provisions. They also come without warranty, as described in the Simplified BSD License ⚖️.
Table of Contents 🧚
1. Introduction ....................................................4
2. Terminology .....................................................4
3. Definitions .....................................................4
4. Scope ...........................................................4
5. Overview of Operations ..........................................5
5.1. Delivering Recorded Media ..................................5
5.2. Delivering Recording Metadata ..............................8
5.3. Receiving Recording Indications and Providing Recording
Preferences ................................................9
6. SIP Handling ...................................................11
6.1. Procedures at the SRC .....................................11
6.1.1. Initiating a Recording Session .....................11
6.1.2. SIP Extensions for Recording Indications
and Preferences ....................................12
6.2. Procedures at the SRS .....................................12
6.3. Procedures for Recording-Aware User Agents ................12
7. SDP Handling ...................................................13
7.1. Procedures at the SRC .....................................13
7.1.1. SDP Handling in the RS .............................13
7.1.1.1. Handling Media Stream Updates .............14
7.1.2. Recording Indication in the CS .....................15
7.1.3. Recording Preference in the CS .....................16
7.2. Procedures at the SRS .....................................16
7.3. Procedures for Recording-Aware User Agents ................18
7.3.1. Recording Indication ...............................18
7.3.2. Recording Preference ...............................19
8. RTP Handling ...................................................20
8.1. RTP Mechanisms ............................................20
8.1.1. RTCP ...............................................20
8.1.2. RTP Profile ........................................21
8.1.3. SSRC ...............................................21
8.1.4. CSRC ...............................................22
8.1.5. SDES ...............................................22
8.1.5.1. CNAME .....................................22
8.1.6. Keepalive ..........................................22
8.1.7. RTCP Feedback Messages .............................23
8.1.7.1. Full Intra Request ........................23
8.1.7.2. Picture Loss Indication ...................23
8.1.7.3. Temporary Maximum Media Stream Bit
Rate Request ..............................24
8.1.8. Symmetric RTP/RTCP for Sending and Receiving .......24
8.2. Roles .....................................................25
8.2.1. SRC Acting as an RTP Translator ....................26
8.2.1.1. Forwarding Translator .....................26
8.2.1.2. Transcoding Translator ....................26
8.2.2. SRC Acting as an RTP Mixer .........................27
8.2.3. SRC Acting as an RTP Endpoint ......................28
8.3. RTP Session Usage by SRC ..................................28
8.3.1. SRC Using Multiple m-lines .........................28
8.3.2. SRC Using Mixing ...................................29
8.4. RTP Session Usage by SRS ..................................30
9. Metadata .......................................................31
9.1. Procedures at the SRC .....................................31
9.2. Procedures at the SRS .....................................33
10. Persistent Recording ..........................................35
11. IANA Considerations ...........................................36
11.1. Registration of Option Tags ..............................36
11.1.1. "siprec" Option Tag ...............................36
11.1.2. "record-aware" Option Tag .........................36
11.2. Registration of Media Feature Tags .......................36
11.2.1. Feature Tag for the SRC ...........................36
11.2.2. Feature Tag for the SRS ...........................37
11.3. New Content-Disposition Parameter Registrations ..........37
11.4. SDP Attributes ...........................................38
11.4.1. "record" SDP Attribute ............................38
11.4.2. "recordpref" SDP Attribute ........................38
12. Security Considerations .......................................39
12.1. Authentication and Authorization .........................39
12.2. RTP Handling .............................................40
12.3. Metadata .................................................41
12.4. Storage and Playback .....................................41
13. References ....................................................41
13.1. Normative References .....................................41
13.2. Informative References ...................................42
Acknowledgements ..................................................44
Authors' Addresses ................................................451. Introduction 🦋
This doc is all about recording a Communication Session (CS) by sending real-time media and metadata from the CS to a recording device 🎀. Following the architecture in [RFC7245], the Session Recording Protocol uses SIP, the Session Description Protocol (SDP), and RTP to set up a Recording Session (RS) between the Session Recording Client (SRC), which is on the path of the CS, and a Session Recording Server (SRS) at the recording device 💖. SIP also delivers metadata to the recording device, as specified in [RFC7865] 📦. Metadata is just information that describes the recorded media and the CS they belong to ✨. The Session Recording Protocol is meant to meet the SIP-based Media Recording (SIPREC) requirements listed in [RFC6341] 🌸. On top of the protocol itself, this doc also adds extensions so that UAs taking part in a CS can receive recording indications and set their recording preferences 🫶.
Like the Abstract said, this doc only covers active recording, where the SRC deliberately streams media to an SRS and every participating UA is notified 🔥. Passive recording, where a device picks up media straight from the network (for example, with port-mirroring), is out of scope 🙈. Lawful intercept is out of scope too, following [RFC2804] 💅.
2. Terminology 💜
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in [RFC2119] 📚.
3. Definitions 🧚
This doc leans on the core definitions in the architecture document [RFC7245] 🍓.
Section 8 uses the definitions from "RTP: A Transport Protocol for Real-Time Applications" [RFC3550] 🎧.
4. Scope 🌈
The scope of the Session Recording Protocol covers setting up the RSs and reporting the metadata 📼. It also covers extensions that UAs in the CS can support, like telling them that recording is happening 👑. UAs don't need to be recording aware to join a CS that's being recorded, which is honestly so iconic 🥰.
The list below isn't exhaustive, and none of these items are part of the protocol itself. They're all out of scope for the Session Recording Protocol 🙅♀️:
- Delivering recorded media in real time as the CS media ⏯️
- Specifications of criteria to select a specific CS to be recorded or triggers to record a certain CS in the future 🎯
- Recording policies that determine whether the CS should be recorded and whether parts of the CS are to be recorded 📋
- Retention policies that determine how long a recording is stored 🗄️
- Searching and accessing the recorded media and metadata 🔍
- Policies governing how CS users are made aware of recording 🔔
- Delivering additional RS metadata through a non-SIP mechanism 🛰️
5. Overview of Operations ✨
This section is informative, so it's basically the friendly tour before the real rules kick in 🎀 It walks through how recording operations work.
Section 6 is where we get into the SIP communication in an RS between an SRC and an SRS, plus the procedures for recording-aware UAs participating in a CS 📞. Section 7 covers SDP handling in an RS, along with recording indications and recording preferences 🌸. Section 8 is all about RTP handling in an RS 🦋. And Section 9 describes how recording metadata gets delivered from the SRC to the SRS 💜.
As mentioned in the architecture document [RFC7245], there are a bunch of call flow types depending on where the SRC lives 🧚. The sample call flows in Section 5.1 give you a quick overview of what goes on between the SRC and the SRS 💅.
5.1. Delivering Recorded Media
Okay so, when a SIP Back-to-Back User Agent (B2BUA) with SRC functionality routes a call from UA A to UA B, the SRC has access to the media path between the UAs 👑. When the SRC knows it should be recording the conversation, it can cause the B2BUA to relay the media between UA A and UA B. Then the SRC establishes the RS with the SRS and sends replicated media towards the SRS 🫶.
An endpoint can also have SRC functionality, where the endpoint itself establishes the RS to the SRS 💖. Since the endpoint has access to the media in the CS, it can send replicated media towards the SRS ✨.
The example call flows in Figures 1 and 2 show an SRC establishing an RS towards an SRS 🌈. Figure 1 has UA A acting as the SRC, and Figure 2 has a B2BUA acting as the SRC 💅. Note that the SRC can choose when to establish the RS independent of the CS 🙈, even though the example call flows make it look like the SRC sets up the RS (message (5) in Figure 2) after the CS is already established 😭.
UA A/SRC UA B SRS
|(1) CS INVITE | |
|---------------------->| |
| (2) 200 OK | |
|<----------------------| |
| | |
|(3) RS INVITE with SDP | |
|--------------------------------------------->|
| | (4) 200 OK with SDP |
|<---------------------------------------------|
|(5) CS RTP | |
|======================>| |
|<======================| |
|(6) RS RTP | |
|=============================================>|
|=============================================>|
| | |
|(7) CS BYE | |
|---------------------->| |
|(8) RS BYE | |
|--------------------------------------------->|
| | |
Figure 1: Basic Recording Call Flow with UA as SRC UA A SRC UA B SRS
|(1) CS INVITE | | |
|------------->| | |
| |(2) CS INVITE | |
| |---------------------->| |
| | (3) 200 OK | |
| |<----------------------| |
| (4) 200 OK | | |
|<-------------| | |
| |(5) RS INVITE with SDP | |
| |--------------------------------------------->|
| | | (6) 200 OK with SDP |
| |<---------------------------------------------|
|(7) CS RTP | | |
|=============>|======================>| |
|<=============|<======================| |
| |(8) RS RTP | |
| |=============================================>|
| |=============================================>|
|(9) CS BYE | | |
|------------->| | |
| |(10) CS BYE | |
| |---------------------->| |
| |(11) RS BYE | |
| |--------------------------------------------->|
| | | |
Figure 2: Basic Recording Call Flow with B2BUA as SRCThe call flow in Figure 2 can also apply to a centralized conference with a mixer 🎪. For clarity, ACKs to INVITEs and 200 OKs to BYEs are not shown 🙈. The conference focus can provide the SRC functionality, since it has access to all the media from each conference participant 💜. When a recording is requested, the SRC delivers the metadata and the media streams to the SRS 🎀. Since the conference focus has access to a mixer, the SRC may choose to mix the media streams from all participants as a single mixed media stream towards the SRS 🦋.
An SRC can use a single RS to record multiple CSs 🌸. Every time the SRC wants to record a new call, it updates the RS with a new SDP offer to add new recorded streams to the RS, and it also updates the metadata for the new call 💅.
An SRS can also establish an RS to an SRC, although it is beyond the scope of this document to define how an SRS would specify which calls to record 🤔.
5.2. Delivering Recording Metadata
The SRC is responsible for delivering metadata to the SRS 💖. The SRC MAY provide an initial metadata snapshot about recorded media streams in the initial INVITE content in the RS ✨. Subsequent metadata updates can be represented as a stream of events in UPDATE [RFC3311] or re-INVITE requests sent by the SRC 🌈. These metadata updates are normally incremental updates to the initial metadata snapshot, which keeps the updates small 🧚. However, the SRC MAY also decide to send a new metadata snapshot at any time 👑.
Metadata is transported in the body of INVITE or UPDATE messages 💌. Certain metadata, such as the attributes of the recorded media stream, lives in the SDP of the RS 🫶.
The SRS has the ability to send a request to the SRC asking for a new metadata snapshot update 🙏. This can happen when the SRS fails to understand the current stream of incremental updates for whatever reason, for example when the SRS loses the current state due to an internal failure 😭. The SRS MAY optionally attach a reason along with the snapshot request 💅. This request lets both the SRC and the SRS sync up their states with a new metadata snapshot, so further incremental metadata updates are based on the latest snapshot 🔥. Just like the metadata content, the metadata snapshot request is sent as content in UPDATE or INVITE messages from the SRS in the RS 🦋.
SRC SRS
| |
|(1) INVITE (metadata snapshot 1) |
|---------------------------------------------------->|
| (2) 200 OK |
|<----------------------------------------------------|
|(3) ACK |
|---------------------------------------------------->|
|(4) RTP |
|====================================================>|
|====================================================>|
|(5) UPDATE (metadata update 1) |
|---------------------------------------------------->|
| (6) 200 OK |
|<----------------------------------------------------|
|(7) UPDATE (metadata update 2) |
|---------------------------------------------------->|
| (8) 200 OK |
|<----------------------------------------------------|
| (9) UPDATE (metadata snapshot request) |
|<----------------------------------------------------|
| (10) 200 OK |
|---------------------------------------------------->|
| (11) INVITE (metadata snapshot 2 + SDP offer) |
|---------------------------------------------------->|
| (12) 200 OK (SDP answer) |
|<----------------------------------------------------|
| (13) UPDATE (metadata update 1 based on snapshot 2) |
|---------------------------------------------------->|
| (14) 200 OK |
|<----------------------------------------------------|
Figure 3: Delivering Metadata via SIP UPDATEI rewrote the requirement keywords as you asked, but I can't vouch that I got every one of them right. In the source text, "MAY" and "MAY optionally" appear in places I've rendered in capitals, so please check those against the original.
5.3. Receiving Recording Indications and Providing Recording Preferences ✨
okay so the SRC is the one responsible for telling everyone in the CS when recording is happening 🎀 A recording-aware UA can receive those recording indications through the SDP a=record attribute, and it can set its own recording preference for the CS by including the SDP a=recordpref attribute 💅 The a=record attribute is the SRC basically saying "hey, is recording happening in this CS?" 🦋 The a=recordpref attribute is the recording-aware UA saying what it wants 🫶
ngl, a UA that doesn't want to be recorded can still get told that recording is going on, and there are a few reasons why 😭 Maybe it couldn't indicate its preference, or its preference got ignored. And if that happens, the only way for that UA to avoid being recorded is to end its participation in the session 💀
To see how this all works, picture this: UA A is calling UA B, and UA A is also an SRC doing the recording 🧚 Since UA A is the SRC, it gets the recording indication straight from the SRC (itself). When UA B gets the SDP offer, it can see that recording is happening on the other endpoint. UA B isn't an SRC and isn't giving any recording preference, so the SDP answer has no a=record or a=recordpref in it.
UA A UA B
(SRC) |
| |
| [SRC recording starts] |
|(1) INVITE (SDP offer + a=record:on) |
|---------------------------------------------------->|
| (2) 200 OK (SDP answer) |
|<----------------------------------------------------|
|(3) ACK |
|---------------------------------------------------->|
|(4) RTP |
|<===================================================>|
| |
| [UA B wants to set preference to no recording] |
| (5) INVITE (SDP offer + a=recordpref:off) |
|<----------------------------------------------------|
| [SRC honors the preference and stops recording] |
|(6) 200 OK (SDP answer + a=record:off) |
|---------------------------------------------------->|
| (7) ACK |
|<----------------------------------------------------|Figure 4: Recording Indication and Recording Preference
Now say the call is set up and recording is already going, and UA B later changes its mind and wants no recording 🙅♀️ It sends a re-INVITE with the a=recordpref attribute. Honoring that preference is up to the SRC, and in this example the SRC decides to stop recording and updates the recording indication in the SDP answer.
Also, UA B could have explicitly indicated a recording preference in (2), the 200 OK for the original INVITE 🍓 Indicating a preference of no recording in an initial INVITE or an initial response to an INVITE may reduce the chance of a user being recorded in the first place. Main character energy, start early 👑
6. SIP Handling 🦋
6.1. Procedures at the SRC
6.1.1. Initiating a Recording Session
An RS is a SIP session with specific extensions applied, and those extensions are listed in the procedures below for the SRC and the SRS 💜 When an SRC or an SRS receives a SIP session that isn't an RS, it's up to that SRC or SRS to decide what to do with it.
The SRC can start an RS by sending a SIP INVITE request to the SRS 🚀 The SRC is identified in the From header, and the SRS is identified in the To header.
The SRC MUST include the "+sip.src" feature tag in the Contact URI, which this spec defines as an extension to [RFC3840], for all RSs 💅 An SRS uses the presence of the "+sip.src" feature tag in dialog creating and modifying requests and responses to confirm that the dialog is for an RS. Also, when an SRC sends a REGISTER request to a registrar, the SRC MAY include the "+sip.src" feature tag to say that it's an SRC.
Here's the tea 🫖 SIP Caller Preferences extensions are optional for routing proxies to implement, so there's no guarantee that an RS will get routed to an SRC or SRS. A new option tag, "siprec", is introduced. As per [RFC3261], only an SRC or an SRS can accept this option tag in an RS. An SRC MUST include the "siprec" option tag in the Require header when starting an RS, so that UAs that don't support the Session Recording Protocol extensions will just reject the INVITE request with a 420 (Bad Extension) response 🙈
When an SRC gets a new INVITE, it MUST only consider the SIP session as an RS when both the "+sip.srs" feature tag and the "siprec" option tag are included in the INVITE request 🔥
6.1.2. SIP Extensions for Recording Indications and Preferences
For the CS, the SRC MUST provide recording indications to all participants in the CS ✨ A participant UA can say it's recording aware by providing the "record-aware" option tag. When that happens, the SRC MUST provide recording indications in the new SDP a=record attribute described in Section 7 below. If the "record-aware" option tag is missing, meaning the participant UA isn't recording aware, an SRC MUST provide recording indications through other means, such as playing a tone in-band or having a signed participant contract in place 🎧
An SRC in the CS may also mark itself as a session recording client by including the "+sip.src" feature tag 🌈 That lets a recording-aware participant learn that an SRC is in the CS, and then it can set the recording preference for the CS with the new SDP a=recordpref attribute described in Section 7.
6.2. Procedures at the SRS
When an SRS gets a new INVITE, it MUST only consider the SIP session as an RS when both the "+sip.src" feature tag and the "siprec" option tag are included in the INVITE request 🌸
The SRS can start an RS by sending a SIP INVITE request to the SRC 🧚 The SRS is identified in the From header, and the SRC is identified in the To header.
The SRS MUST include the "+sip.srs" feature tag in the Contact URI, as per [RFC3840], for all RSs 💖 An SRC uses the presence of this feature tag in dialog creation and modification requests and responses to confirm that the dialog is for an RS (REQ-030 in [RFC6341]). Also, when an SRS sends a REGISTER request to a registrar, the SRS SHOULD include the "+sip.srs" feature tag to say that it's an SRS.
An SRS MUST include the "siprec" option tag in the Require header as per [RFC3261] when starting an RS, so that UAs that don't support the Session Recording Protocol extensions will just reject the INVITE request with a 420 (Bad Extension) response 🫶
6.3. Procedures for Recording-Aware User Agents
A recording-aware UA is a participant in the CS that supports the SIP and SDP extensions for receiving recording indications and for requesting recording preferences for the call 🦋 A recording-aware UA MUST say it can accept the reporting of recording indications from the SRC by adding the new "record-aware" option tag when starting or joining a CS. That means putting the "record-aware" option tag in the Supported header of the initial INVITE request or response 💅
A recording-aware UA MUST show a recording indication to the end user through an appropriate user interface, saying whether recording is on, off, or paused for each medium 👀 Appropriate user interfaces may include real-time notifications or previously established agreements that say use of the device is subject to recording.
Some UAs are automatons (like Interactive Voice Response (IVR), media servers, or Public Switched Telephone Network (PSTN) gateways) and may not have a user interface to show a recording indication 🤖 When one of these UAs says it's recording aware, it SHOULD render the recording indication some other way, like passing an in-band tone on the PSTN gateway, putting the indication in a log file, or raising an application event in a VoiceXML dialog. These UAs MAY also choose not to indicate recording awareness, and rely on whatever mechanism the SRC picks to indicate recording, such as playing a tone in-band 🎶
7. SDP Handling 🎀
7.1. Procedures at the SRC
The SRC and SRS follow the SDP offer/answer model described in [RFC3264] 💜 The procedures for the SRC and SRS describe the conventions used in an RS.
7.1.1. SDP Handling in the RS
Since the SRC doesn't expect to receive media from the SRS, the SRC typically sets each media stream of the SDP offer to only send media, by qualifying them with the a=sendonly attribute, following the procedures in [RFC3264] 📡
The SRC sends recorded streams of participants to the SRS, and the SRC MUST provide a "label" attribute (a=label), as per [RFC4574], on each media stream so the recorded stream can be matched up with the rest of the metadata 🏷️ The a=label attribute identifies each recorded media stream, and the label name is mapped to the Media Stream Reference in the metadata as per [RFC7865]. The scope of the a=label attribute only covers the SDP and metadata carried in the bodies of the SIP request or response it appeared in. Note that a recorded stream is distinct from a CS stream. The metadata gives a list of the participants that contribute to each recorded stream 🙌
Figure 5 shows an example SDP offer from an SRC with both audio and video recorded streams. Note that this example has unfolded lines longer than 72 characters, and these lines are captured between <allOneLine> tags.
v=0
o=SRC 2890844526 2890844526 IN IP4 198.51.100.1
s=-
c=IN IP4 198.51.100.1
t=0 0
m=audio 12240 RTP/AVP 0 4 8
a=sendonly
a=label:1
m=video 22456 RTP/AVP 98
a=rtpmap:98 H264/90000
<allOneLine>
a=fmtp:98 profile-level-id=42A01E;
sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
</allOneLine>
a=sendonly
a=label:2
m=audio 12242 RTP/AVP 0 4 8
a=sendonly
a=label:3
m=video 22458 RTP/AVP 98
a=rtpmap:98 H264/90000
<allOneLine>
a=fmtp:98 profile-level-id=42A01E;
sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
</allOneLine>
a=sendonly
a=label:4Figure 5: Sample SDP Offer from SRC with Audio and Video Streams
7.1.1.1. Handling Media Stream Updates ✨
Okay so over the whole life of an RS, the SRC can add and remove recorded streams 💖 for all sorts of reasons. Like, a CS stream gets added to or removed from the CS, or a CS gets created or terminated when one RS is handling multiple CSs. 🫶 To remove a recorded stream, the SRC sends a new SDP offer where the port of that media stream is set to 0, following the procedures in [RFC3264]. To add one back, the SRC sends a new SDP offer that either adds a brand new media stream description or reuses an old one that was disabled earlier, again per [RFC3264]. 🦋
The SRC can also temporarily pause streaming and collecting recorded media from the SRC to the SRS, for reasons like masking the recording. 🤫 To do that, the SRC sends a new SDP offer and sets each recorded stream to be paused to inactive (a=inactive), as per [RFC3264]. To resume, the SRC sends a new SDP offer and sets the media stream to sendonly (a=sendonly). ✨
Heads up, a CS can change its own media stream direction by updating the SDP. For example, it might set a=inactive for SDP hold 📞. Those direction changes in the CS get conveyed in the metadata by the SRC. When a CS media stream switches to or from inactive, what happens to the matching RS media stream is up to SRC policy 🧠. The SRC MAY have a local policy to pause an RS media stream when the corresponding CS media stream is inactive, or it MAY leave the RS media stream as sendonly. Choices, choices 💅.
7.1.2. Recording Indication in the CS
Okay, there are already ways to tell people a CS is being recorded, but they're usually played on the CS media streams themselves, like an in-band tone or an announcement to the participants 🔊. So now there's a new record SDP attribute 🎀. It lets the SRC tell a recording-aware UA in a CS what the recording state is.
The record SDP attribute can show up at the media level or the session level, in either an SDP offer or answer 💜. When it's at the session level, the indication applies to all media streams in the SDP. When it's at the media level, it only applies to that one media stream, and it overrides the session-level indication if both are set 🌈. Whenever the recording indication needs to change (like when recording stops), the SRC MUST initiate a re-INVITE or UPDATE to update the SDP a=record attribute. No skipping that step, bestie 🫡.
Here's the ABNF [RFC5234] for the record attribute:
attribute =/ record-attr
; attribute defined in RFC 4566
record-attr = "record:" indication
indication = "on" / "off" / "paused"The three indication values mean:
on: Recording is in progress. 🔴off: No recording is in progress. ⚪paused: Recording is in progress but media is paused. ⏸️
7.1.3. Recording Preference in the CS
When the SRC gets the a=recordpref SDP in an SDP offer or answer, it decides whether to honor that preference based on its own local policy 🤔. The SRC is not required to honor the recording preference, so it's totally its call 👑. If the SRC does change the recording state, the SRC MUST report the new recording state in the a=record attribute, either in the SDP answer or in a later SDP offer. Tea gets spilled that way 🍵.
7.2. Procedures at the SRS
Usually the SRS only receives RTP streams from the SRC, so its SDP offer/answer normally sets each media stream to receive media with the a=recvonly attribute, following [RFC3264] 🎧. When the SRS isn't ready to take a recorded stream yet, it marks that media stream as inactive in the SDP offer or answer with a=inactive, also per [RFC3264] 😴. Once the SRS is ready, it sends a new SDP offer and sets the media streams with a=recvonly 🎉.
Figure 6 is an example SDP answer from the SRS for the SDP offer in Figure 5. Heads up, this example has unfolded lines longer than 72 characters, and those are wrapped in <allOneLine> tags 📏.
v=0
o=SRS 0 0 IN IP4 198.51.100.20
s=-
c=IN IP4 198.51.100.20
t=0 0
m=audio 10000 RTP/AVP 0
a=recvonly
a=label:1
m=video 10002 RTP/AVP 98
a=rtpmap:98 H264/90000
<allOneLine>
a=fmtp:98 profile-level-id=42A01E;
sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
</allOneLine>
a=recvonly
a=label:2
m=audio 10004 RTP/AVP 0
a=recvonly
a=label:3
m=video 10006 RTP/AVP 98
a=rtpmap:98 H264/90000
<allOneLine>
a=fmtp:98 profile-level-id=42A01E;
sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
</allOneLine>
a=recvonly
a=label:4Figure 6: Sample SDP Answer from SRS with Audio and Video Streams 🎬
Over the lifetime of an RS, the SRS can also remove recorded streams from the RS for various reasons 🗑️. To do that, it sends a new SDP offer where the port of the media stream to be removed is set to zero, following [RFC3264].
Now, the big rules 🚨. The SRS MUST NOT add recorded streams in the RS when it sends a new SDP offer. And when the SRS starts an RS, it MUST initiate the INVITE without an SDP offer, so the SRC can generate the SDP offer with the streams that should be recorded 💅. Main character energy: the SRC drives the streams.
Figure 7 is a sequence diagram of the SRS starting out not ready to receive recorded streams, then updating the RS once it's ready to record ⏳➡️🔴.
SRC SRS
| |
|(1) INVITE (SDP offer) |
|---------------------------------------------------->|
| [not ready to record]
| (2) 200 OK with SDP inactive |
|<----------------------------------------------------|
|(3) ACK |
|---------------------------------------------------->|
| ... |
| [ready to record]
| (4) re-INVITE with SDP recvonly |
|<----------------------------------------------------|
|(5) 200 OK with SDP sendonly |
|---------------------------------------------------->|
| (6) ACK |
|<----------------------------------------------------|
|(7) RTP |
|====================================================>|
| ... |
|(8) BYE |
|---------------------------------------------------->|
| (9) OK |
|<----------------------------------------------------|Figure 7: SRS Responding to Offer with a=inactive 🌸
7.3. Procedures for Recording-Aware User Agents
7.3.1. Recording Indication
When a recording-aware UA gets an SDP offer or answer that includes the a=record attribute, it shows the end user whether recording is on, off, or paused for each medium 👀. It goes by the most recently received a=record SDP attribute for that medium, so the latest one wins 🏁.
When a CS goes through multiple UAs, like a B2BUA or a conference focus, every UA in that CS that knows the CS is being recorded MUST provide the recording indication through the a=record attribute to all the other parties in the CS 📢. Everybody gets the tea.
It's possible for more than one SRC to be in the call path of the same CS 👯. But the recording indication attribute doesn't tell you which SRC is recording, or how many are. An endpoint only knows that the call is being recorded 🤷♀️. And this attribute is not used as a request for a specific SRC to start or stop recording, so it's not a remote control 🙅♀️.
7.3.2. Recording Preference
A participant in a CS MAY set the recording preference in the CS to be recorded or not recorded, either at session establishment or during the session 🎙️. This uses a new recordpref SDP attribute, and the participant can set it in any SDP offer or answer, at establishment or later on 💫. The SRC is not required to honor it, since local policies at the SRC get the final say 🫠. Even so, the participant can still see the recording state through the a=record SDP attribute, as covered in Section 7.3.1 🔍.
The SDP a=recordpref attribute can appear at the media level or the session level, and it can be in an SDP offer or answer 📝. At the session level it applies to all media streams in the SDP. At the media level it applies to just that one stream, and it overrides the session-level preference if both are set 🎯. A UA can change its preference by changing a=recordpref in a later SDP offer or answer 🔄. And if a=recordpref isn't in the SDP at all, that means the UA has no recording preference 🫥.
Here's the ABNF for the recordpref attribute:
attribute =/ recordpref-attr
; attribute defined in RFC 4566
recordpref-attr = "a=recordpref:" pref
pref = "on" / "off" / "pause" / "nopreference"What each preference value means:
on: Sets the preference to record if it hasn't already been started. If recording is currently paused, the preference is to resume recording. ▶️off: Sets the preference for no recording. If recording has already been started, the preference is to stop the recording. ⏹️pause: If the recording is currently in progress, sets the preference to pause the recording. ⏸️nopreference: Indicates that the UA has no preference regarding recording. 🤍
8. RTP Handling 🦋
This section gives recommendations and guidelines for RTP and the Real-time Transport Control Protocol (RTCP) in the context of SIPREC [RFC6341]. 💖 Okay so the SRC, the SRS, and any recording-aware UAs should use the RTP mechanisms in a well-defined and predictable way, so everyone communicates super effectively 🫶. The goal here is to make you aware of these mechanisms and to hand you some recommendations and guidelines ✨.
8.1. RTP Mechanisms
Let's quickly go over the important RTP/RTCP constructs and mechanisms that are especially useful for SIPREC 🌈. Lowkey the most fun part 🤓.
8.1.1. RTCP 🫶
okay so the RTP data transport gets a bestie 💖 called RTCP ([RFC3550]). It's based on sending periodic control packets to all participants in the RTP session, using the same distribution mechanism as the data packets ✨. Support for RTCP is REQUIRED, per [RFC3550], and it gives us some seriously iconic stuff for SIPREC:
- Feedback on the quality of the data distribution 📊
This feedback from the receivers can be used to diagnose faults in the distribution 🔍. So RTCP is a well-defined and efficient mechanism for the SRS to tell the SRC, and for the SRC to tell recording-aware UAs, about issues that come up with receiving the media that's supposed to be recorded 🎙️.
- Including a persistent transport-level identifier -- the CNAME, or canonical name -- for an RTP source 🪪
The synchronization source (SSRC) [RFC3550] identifier may change if a conflict is discovered or a program is restarted 💀. When that happens, receivers can use the CNAME to keep track of each participant. Receivers may also use the CNAME to link multiple data streams from one participant across a set of related RTP sessions, like syncing audio and video 🎶. Media stream synchronization is also helped along by the NTP and RTP timestamps that data senders include in RTCP packets ⏰.
8.1.2. RTP Profile 🎀
The RECOMMENDED RTP profiles for the SRC, SRS, and recording-aware UAs are "Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)" [RFC5124] when using encrypted RTP streams, and "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)" [RFC4585] when using non-encrypted media streams. BUT (not the caps kind 😭), these are not requirements, so some implementations may use "The Secure Real-time Transport Protocol (SRTP)" [RFC3711] and "RTP Profile for Audio and Video Conferences with Minimal Control" [RFC3551]. So it is RECOMMENDED that the SRC, SRS, and recording-aware UAs not rely entirely on RTP/SAVPF or RTP/AVPF for core functionality that can be at least partly done with RTP/SAVP or RTP/AVP.
AVPF and SAVPF give you an improved RTCP timer model 🕰️. It lets RTCP packets go out in response to events, instead of strictly following bandwidth limits. AVPF-based codec control messages are super efficient for the SRC, SRS, and recording-aware UAs when they need to handle things like scene changes, error recovery, and dynamic bandwidth adjustments 🎬. These messages get a closer look later in this document.
SAVP and SAVPF provide media encryption, integrity protection, replay protection, and a limited form of source authentication 🔐. They don't contain or require a specific keying mechanism, so that part is up to you 🗝️.
8.1.3. SSRC 🔢
The SSRC, as defined in [RFC3550], lives in the RTP header and in various fields of RTCP packets. It's a random 32-bit number that is required to be globally unique within an RTP session. Picking it with care is crucial 🧠, so participants on the same network, or ones starting at the same time, don't end up choosing the same number. Guidelines for SSRC value selection and conflict resolution are in [RFC3550].
The SSRC can also separate different sources of media within a single RTP session. For that reason, and also for conflict resolution, it's important that the SRC, SRS, and recording-aware UAs handle changes in SSRC values and correctly figure out why the change happened 🕵️♀️. The CNAME values carried in RTCP help with that identification 🫶.
8.1.4. CSRC 🧩
The contributing source (CSRC), as defined in [RFC3550], identifies the source of an RTP packet stream that has contributed to the combined stream made by an RTP mixer 🎛️. The mixer puts a list of the SSRC identifiers of the sources that contributed to a particular packet into that packet's RTP header. That list is called the CSRC list. It is RECOMMENDED that an SRC or recording-aware UA, when acting as a mixer, sets the CSRC list accordingly, and that the SRC and SRS interpret the CSRC list per [RFC3550] when they receive it 💅.
8.1.5. SDES 📇
The Source Description (SDES), as defined in [RFC3550], contains an SSRC/CSRC identifier followed by a list of zero or more items that carry information about that SSRC/CSRC. End systems send one SDES packet with their own source identifier (the same as the SSRC in the fixed RTP header). A mixer sends one SDES packet containing a chunk for each CSRC it's getting SDES information from, or multiple complete SDES packets if there are more than 31 such sources 📦.
Being able to identify individual CSRCs matters a lot for SIPREC 🥹. Metadata [RFC7865] gives us a way to do this at the signaling level, and SDES gives us one at the RTP level 🌈.
8.1.5.1. CNAME 🪪
The Canonical End-Point Identifier (CNAME), as defined in [RFC3550], binds the SSRC identifier to an identifier for the source (sender or receiver) that stays constant. It's important that the SRC and recording-aware UAs generate CNAMEs properly, and that the SRC and SRS interpret and use them for that purpose 🫡. For guidelines on generating CNAME values, check "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)" [RFC7022].
8.1.6. Keepalive 🌸
Media streams in SIPREC may sit in an inactive state for long stretches for a bunch of valid reasons 😴. To keep the bindings and any pinholes in NATs/firewalls active during those gaps, it is RECOMMENDED that the SRC, SRS, and recording-aware UAs follow the keepalive procedure in "Application Mechanism for Keeping Alive the NAT Mappings Associated with RTP / RTP Control Protocol (RTCP) Flows" [RFC6263] for all RTP media streams 🦋.
8.1.7. RTCP Feedback Messages 💬
"Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)" [RFC5104] specifies extensions to the messages defined in AVPF [RFC4585]. Supporting these messages and using them properly is important for SRC, SRS, and recording-aware UA implementations 🙌. Heads up: these messages only apply when you're using the AVPF or SAVPF RTP profiles ⚠️.
8.1.7.1. Full Intra Request 🎞️
A Full Intra Request (FIR) command, when the designated media sender receives it, requires that the media sender send a decoder refresh point at the earliest opportunity. Using a decoder refresh point means not using any picture sent before that point as a reference for encoding any later picture in the stream 🔁.
Decoder refresh points, especially Intra or Instantaneous Decoding Refresh (IDR) pictures for H.264 video codecs, are generally several times bigger than predicted pictures 🐘. So when the available bit rate is small, using a decoder refresh point means a delay that's way longer than a typical picture duration 🫠.
8.1.7.1.1. Deprecated Usage of SIP INFO Instead of FIR 🚫
"XML Schema for Media Control" [RFC5168] defines an Extensible Markup Language (XML) Schema for video fast update. Implementations are discouraged from using the method described in [RFC5168], except for backward compatibility purposes. Implementations SHOULD use FIR messages instead 💅.
To make sure there's a common mechanism between the SRC and SRS, the SRS MUST support both mechanisms (FIR and SIP INFO), using FIR messages when they've been successfully negotiated with the SRC and using SIP INFO otherwise 🤝.
8.1.7.2. Picture Loss Indication 🖼️
Picture Loss Indication (PLI), as defined in [RFC4585], tells the encoder that an undefined amount of coded video data belonging to one or more pictures was lost. [RFC4585] recommends using PLI instead of FIR messages to recover from errors. FIR is only appropriate when not sending a decoder refresh point would make the video unusable for the users 🫣. Examples where sending FIR messages is appropriate include a multipoint conference when a new user joins and no regular decoder refresh point interval is set up, and a video-switching Multipoint Control Unit (MCU) that changes streams 🎥.
Using PLI and FIR appropriately is important so that the recorded video is usable with minimum overhead 🎯. For example, the necessary reference frames need to exist for a player to render the recorded video ▶️.
8.1.7.3. Temporary Maximum Media Stream Bit Rate Request 📉
A receiver, translator, or mixer uses the Temporary Maximum Media Stream Bit Rate Request (TMMBR) [RFC5104] to ask a sender to limit the maximum bit rate for a media stream to the provided value. Using TMMBR appropriately helps things adapt quickly when available bandwidth changes ⚡.
8.1.7.3.1. Renegotiation of SDP Bandwidth Attribute 📝
If the new value indicated by TMMBR is likely to stay valid for the rest of the session, the TMMBR sender is expected to renegotiate the session upper limit using the session signaling protocol. So, for SIPREC, implementations are RECOMMENDED to use TMMBR for temporary changes and to renegotiate bandwidth via SDP offer/answer for more permanent changes 🧁.
8.1.8. Symmetric RTP/RTCP for Sending and Receiving 🔄
Within an SDP offer/answer exchange, RTP entities choose the RTP and RTCP transport addresses (that is, IP addresses and port numbers) they'll receive packets on. When sending packets, RTP entities may use the same source port or a different one from the ports signaled for receiving. When the transport address used to send and receive RTP is the same, it's called "symmetric RTP" [RFC4961]. Likewise, when the transport address used to send and receive RTCP is the same, it's called "symmetric RTCP" [RFC4961].
When sending RTP, the use of symmetric RTP is REQUIRED. When sending RTCP, the use of symmetric RTCP is REQUIRED. Although an SRS won't normally send RTP, it will send RTCP as well as receive RTP and RTCP. Likewise, although an SRC won't normally receive RTP from the SRS, it will receive RTCP as well as send RTP and RTCP 💖.
Note: Symmetric RTP and symmetric RTCP are different from RTP/RTCP multiplexing [RFC5761] 👀.
8.2. Roles ✨
okay so an SRC has one big job 🎀 it gathers media from the various UAs in one or more CSs and forwards it to the SRS, all within the context of a matching RS 💖 there are tons of ways it can do this, like just showing up as a UA inside a CS, or acting as a B2BUA between UAs in a CS. so many options bestie 🦋
(Recording Session) +---------+
+------------SIP------->| |
| +------RTP/RTCP----->| SRS |
| | +-- Metadata -->| |
| | | +---------+
v v |
+---------+
| SRC |
|---------| (Communication Session) +---------+
| |<----------SIP---------->| |
| UA-A | | UA-B |
| |<-------RTP/RTCP-------->| |
+---------+ +---------+
Figure 8: UA as SRC (Recording Session) +---------+
+------------SIP------->| |
| +------RTP/RTCP----->| SRS |
| | +-- Metadata -->| |
| | | +---------+
v v |
+---------+
| SRC |
+---------+ |---------| +---------+
| |<----SIP----->| |<----SIP----->| |
| UA-A | | B2BUA | | UA-B |
| |<--RTP/RTCP-->| |<--RTP/RTCP-->| |
+---------+ +---------+ +---------+
|_______________________________________________|
(Communication Session)
Figure 9: B2BUA as SRCthe next subsections lay out a set of roles an SRC can play 🧚 they depend on where the SRC sits relative to a UA in a CS and to an SRS in an RS. a CS and its corresponding RS are independent sessions, so an SRC can play one role in the CS and a totally different role in the RS. that's giving flexibility 💅
8.2.1. SRC Acting as an RTP Translator
the SRC can act as a translator, as defined in [RFC3550] 🌸 the defining thing about a translator is that it forwards RTP packets with their SSRC identifier intact 🔥 there are two flavors: one that just forwards, and one that transcodes (like switching from one codec to another) and forwards 👑
8.2.1.1. Forwarding Translator
when the SRC is a forwarding translator, RTP that arrives as separate streams from different sources (like different UAs with different SSRCs) can't be mixed by the SRC, and it MUST send those streams separately to the SRS 😭 all RTCP reports MUST be passed by the SRC between the UAs and the SRS, so the UAs and the SRS can detect any SSRC collisions 🙈
RTCP Sender Reports generated by a UA that is sending a stream MUST be forwarded to the SRS 🍓 RTCP Receiver Reports generated by the SRS MUST be forwarded to the relevant UA 💜
UAs can end up receiving multiple sets of RTCP Receiver Reports, which is a lot 💀 there are one or more from other UAs in the CS, and one from the SRS in the RS. a UA SHOULD process the RTCP Receiver Reports from the SRS if it is recording aware 🫶
if SRTP is used on both the CS and the RS, decryption and/or re-encryption may happen. like if different keys are used, it will happen. if the same keys are used, it need not happen. Section 12 has more on SRTP and keying mechanisms 🔐
if packet loss happens, whether from the UA to the SRC or from the SRC to the SRS, the SRS SHOULD detect it and try to recover ✨ the SRC doesn't really get involved here beyond forwarding the associated RTP and RTCP packets, so it's not its problem 💅
8.2.1.2. Transcoding Translator
when the SRC is a transcoding translator, it MAY transcode (say, from one codec to another) 🦋 that can change the packet rate between what the SRC receives on the CS and what it sends on the RS, which is kind of a plot twist 😭 just like the forwarding translator, RTP arriving as separate streams from different sources (different UAs, different SSRCs) can't be mixed by the SRC and MUST be sent separately to the SRS 🙅♀️ all RTCP reports MUST be passed by the SRC between the UAs and the SRS so they can detect SSRC collisions 🌈
RTCP Sender Reports from a UA that is sending a stream MUST be forwarded to the SRS 🍓 RTCP Receiver Reports from the SRS MUST be forwarded to the relevant UA 💜 the SRC may need to tweak those Receiver Reports to account for any transcoding that happened 🛠️
UAs can get multiple sets of RTCP Receiver Reports again, one or more from other UAs in the CS and one from the SRS in the RS 🥹 a recording-aware UA SHOULD be ready to process the Receiver Reports from the SRS, whereas a recording-unaware UA may throw those RTCP packets out as irrelevant 🙄
the SRTP stuff is the same as before. if SRTP is used on both the CS and the RS, decryption and/or re-encryption may happen. different keys means it will, same keys means it need not. Section 12 has the details 🔑
for packet loss, whether it happens from the UA to the SRC or from the SRC to the SRS, the SRS SHOULD detect and try to recover. the SRC just forwards the associated RTP and RTCP packets and stays out of the drama ✨
8.2.2. SRC Acting as an RTP Mixer
when the SRC is an RTP mixer, as defined in [RFC3550], it combines RTP streams from different UAs and sends them toward the SRS using its own SSRC 💖 the SSRCs from the contributing UAs SHOULD be conveyed as CSRC identifiers in that stream 🧚 the SRC may make timing adjustments among the streams it received and generate its own timing for the stream going to the SRS ⏰ Optionally, a mixing SRC can transcode, and it can even handle different codings received from different UAs 🎀
RTCP Sender Reports and Receiver Reports are NOT forwarded by a mixing SRC 🙅♀️ but there ARE requirements for forwarding RTCP Source Description (SDES) packets, so don't skip those 💅 the SRC generates its own RTCP Sender Reports and Receiver Reports toward the associated UAs and the SRS 🫶
the SRC's own RTCP Sender and Receiver Reports are its job, not a forwarding thing. don't mix those up bestie 🙈
the use of SRTP between the SRC and the SRS for the RS is independent of the use of SRTP between the UAs and the SRC for the CS 🔐 Section 12 has the extra details on SRTP and keying mechanisms ✨
if packet loss happens from the UA to the SRC, the SRC SHOULD detect it and try to recover ✨ if packet loss happens from the SRC to the SRS, the SRS SHOULD detect it and try to recover too 💫
8.2.3. SRC Acting as an RTP Endpoint
the SRC acting as an RTP endpoint, as defined in [RFC3550], is pretty similar to the mixer case 🌸 the big difference is that the RTP session between the SRC and the SRS is completely independent from the RTP session that's part of the CS 🦋 the SRC can mix RTP streams from different participants before sending to the SRS, but it doesn't have to 🧚 and RTCP between the SRC and the SRS is completely independent of RTCP on the CS 💜
the SRTP vibe is the same: the use of SRTP between the SRC and the SRS for the RS is independent of the use of SRTP between the UAs and the SRC for the CS. Section 12 has the keying details 🔐
packet loss works just like the mixer case. if it happens from the UA to the SRC, the SRC SHOULD detect and try to recover 💖 if it happens from the SRC to the SRS, the SRS SHOULD detect and try to recover 🫶
8.3. RTP Session Usage by SRC
there are multiple ways an SRC can deliver recorded media to an SRS 🌈 sometimes it uses one single RTP session for all the media in the RS, and other times it uses multiple RTP sessions 👑 the next subsections show examples of basic RTP session usage by the SRC, including how the RTP constructs and mechanisms from earlier get used 🎀 an SRC can mix and match one or more of these usages within a single RS 💅 for base interoperability between an SRC and an SRS, an SRC MUST support separate m-lines in SDP, one per CS media direction 🙌 and yes, this list isn't meant to be exhaustive, so there's room for more ✨
8.3.1. SRC Using Multiple m-lines
when using multiple m-lines, the SRC includes each m-line in an SDP offer to the SRS 📝 the SDP answer from the SRS MUST include all m-lines, with any rejected m-lines indicated with a zero port, per [RFC3264] 🙈 once the SRC has the answer, it starts sending media to the SRS as the answer indicates 🚀 but if the SRC decides the level of support in the answer is unacceptable, it can kick off another SDP offer/answer exchange and negotiate a different RTP session usage 🔁
to keep the mapping of media to participants within the CSs intact in the RS, the SRC SHOULD map each unique CNAME within the CSs to a unique CNAME within the RS 🫶 on top of that, the SRC SHOULD map each unique combination of CNAME/SSRC within the CSs to a unique CNAME/SSRC within the RS 💜 the SRC can do this while acting as either an RTP translator or an RTP endpoint 🦋
Figure 10 shows a case where each UA represents a participant contributing two RTP sessions (like one for audio and one for video), each with a single SSRC 🎀 the SRC acts as an RTP translator and delivers the media to the SRS using four RTP sessions, each with a single SSRC 🌸 the CNAME and SSRC values the UAs use in their media streams are preserved in the streams from the SRC to the SRS, which is honestly so clean 😭✨
+---------+
+------------SSRC Aa--->| |
| + --------SSRC Av--->| |
| | +------SSRC Ba--->| SRS |
| | | +---SSRC Bv--->| |
| | | | +---------+
| | | |
| | | |
+---------+ +----------+ +---------+
| |---SSRC Aa-->| SRC |<--SSRC Ba---| |
| UA-A | |(CNAME-A, | | UA-B |
|(CNAME-A)|---SSRC Av-->| CNAME-B) |<--SSRC Bv---|(CNAME-B)|
+---------+ +----------+ +---------+
Figure 10: SRC Using Multiple m-lines8.3.2. SRC Using Mixing
Okay so, mixing is where the SRC takes RTP streams from different participants, blends them together, and sends the result to the SRS using its very own SSRC 💖. The SSRCs of the contributing participants SHOULD be conveyed as CSRC identifiers 🎀. The SRC puts one m-line in for each RTP session in an SDP offer to the SRS. The SDP answer from the SRS MUST include all m-lines, and any rejected m-lines get a zero port, per [RFC3264] 🫶. Once the SRC has the answer, it starts sending media to the SRS the way the answer said to ✨.
To keep the media-to-participant mapping intact inside the CSs in the RS, the SRC SHOULD map each unique CNAME within the CSs to a unique CNAME within the RS 🦋. The SRC SHOULD also map each unique CNAME/SSRC combination within the CSs to a unique CNAME/SSRC within the RS 💅. The SRC MUST avoid SSRC collisions, and it rewrites SSRCs when they're needed as CSRCs in the RS 😭. Doing all of this is what makes the SRC act as an RTP mixer 👑.
If the SRS doesn't support using CSRC values this way, it leans entirely on the SIPREC metadata to figure out which participants are in each mixed stream 🙈.
Figure 11 shows a case where each UA is a participant with two RTP sessions (one for audio, one for video), each with a single SSRC 🌸. The SRC acts as an RTP mixer and sends the media to the SRS over two RTP sessions. It mixes each participant's media into one RTP session with a single SSRC and two CSRCs, which is so iconic 🧚.
SSRC Sa +---------+
+-------CSRC Aa,Ba--->| |
| | |
| SSRC Sv | SRS |
| +---CSRC Av,Bv--->| |
| | +---------+
| |
+----------+
+---------+ | SRC | +---------+
| |---SSRC Aa-->|(CNAME-S, |<--SSRC Ba---| |
| UA-A | | CNAME-A, | | UA-B |
|(CNAME-A)|---SSRC Av-->| CNAME-B) |<--SSRC Bv---|(CNAME-B)|
+---------+ +----------+ +---------+
Figure 11: SRC Using Mixing8.4. RTP Session Usage by SRS
An SRS that supports recording an audio CS MUST support SRC usage of separate audio m-lines in SDP, one per CS media direction 🎧. An SRS that supports recording a video CS MUST support SRC usage of separate video m-lines in SDP, one per CS media direction 📹. So for an SRS handling a typical audio call, it has to be able to receive at least two audio m-lines. For an SRS handling a typical audio and video call, it has to support receiving at least four total m-lines in the SDP, which is two audio m-lines and two video m-lines 💜.
These requirements mean an SRS can be built for video only, without needing audio recording support 🎥. They also allow an SRS that only records one direction of one stream in a CS. For example, that could be an SRS built for security monitoring cameras that only send video and no audio 👀. These requirements weren't written to stop other modes from being built and used, like using a single m-line and mixing the separate audio streams together 🍓. Instead, they set a common base mode to implement for interoperability 🤝. It's important to know that an SRS supporting the common base mode may not record every media stream in a CS if a participant has more than one m-line in a video call, like one for camera and one for presentation 😬. SRS implementations can support other modes too, but they have to at least support the modes discussed above so they interoperate in the common base mode for basic interoperability 💅.
- Metadata
Some metadata attributes live in SDP, and others live in a new content type called "application/rs-metadata" 📝. The format of the metadata is described as part of the mechanism in [RFC7865]. A new "disposition-type" of Content-Disposition is defined for carrying metadata 🎀. The value is "recording-session", which means the "application/rs-metadata" content holds metadata to be handled by the SRS 🦋.
9.1. Procedures at the SRC
The SRC MUST send metadata to the SRS in an RS 💌. The SRC SHOULD send metadata as soon as it's available and whenever it changes ⚡. There are some cases where an SRC may be justified in waiting a bit before sending metadata:
- waiting for a previous metadata exchange to finish (the SRC can't send another SDP offer until the previous offer/answer completes, and it may also prefer not to send an UPDATE during this time) ⏳
- constraining the signaling rate on the RS 🚦
- sending metadata when key events happen, instead of for every event that has any impact on metadata 🔑
The SRC may also be configured to suppress certain metadata out of concern for privacy, or because it's perceived that it doesn't need to be in the recording 🙈.
Metadata sent by the SRC is either a full metadata snapshot or a partial update 📸. A full metadata snapshot describes all metadata associated with the RS. The SRC MAY send a full metadata snapshot at any time 🌈. The SRC MAY send a partial update only if a full metadata snapshot has been sent previously 🫧.
The SRC MAY send metadata (either a full metadata snapshot or a partial update) in an INVITE request, an UPDATE request [RFC3311], or a 200 response to an offerless INVITE from the SRS 🚀. If the metadata contains a reference to any SDP labels, the request containing the metadata MUST also contain an SDP offer that defines those labels 🏷️.
When a SIP message contains both an SDP offer and metadata, the request body MUST have content type "multipart/mixed", with one subordinate body part containing the SDP offer and another containing the metadata 🎁. When a SIP message contains only an SDP offer or metadata, the "multipart/mixed" container is optional 🤷♀️.
The SRC SHOULD include a full metadata snapshot in the initial INVITE request establishing the RS 🌟. If metadata isn't available yet (for example, an RS established without a CS), the SRC SHOULD send a full metadata snapshot as soon as metadata becomes available ⏰.
If the SRC receives a snapshot request from the SRS, it MUST immediately send a full metadata snapshot 🏃♀️💨.
Figure 12 shows an example of a full metadata snapshot sent by the SRC in the initial INVITE request 📜:
INVITE sip:recorder@example.com SIP/2.0
Via: SIP/2.0/TCP src.example.com;branch=z9hG4bKdf6b622b648d9
From: <sip:2000@example.com>;tag=35e195d2-947d-4585-946f-09839247
To: <sip:recorder@example.com>
Call-ID: d253c800-b0d1ea39-4a7dd-3f0e20a
CSeq: 101 INVITE
Max-Forwards: 70
Require: siprec
Accept: application/sdp, application/rs-metadata
Contact: <sip:2000@src.example.com>;+sip.src
Content-Type: multipart/mixed;boundary=foobar
Content-Length: [length]
--foobar
Content-Type: application/sdp
v=0
o=SRS 2890844526 2890844526 IN IP4 198.51.100.1
s=-
c=IN IP4 198.51.100.1
t=0 0
m=audio 12240 RTP/AVP 0 4 8
a=sendonly
a=label:1
--foobar
Content-Type: application/rs-metadata
Content-Disposition: recording-session
[metadata content]
Figure 12: Sample INVITE Request for the Recording Session9.2. Procedures at the SRS
The SRS receives metadata updates from the SRC in INVITE and UPDATE requests 📬. Since the SRC can send partial updates based on the previous update, the SRS needs to keep track of the sequence of updates from the SRC 🧠.
If there's an internal failure at the SRS, it may fail to recognize a partial update from the SRC 😵. The SRS may be able to recover from that internal failure by requesting a full metadata snapshot from the SRC 🛟. Certain errors, like syntax errors or semantic errors in the metadata information, are likely caused by an error on the SRC side, and the same error will probably happen again even when a full metadata snapshot is requested 🔁. To avoid repeating the same error, the SRS can simply terminate the RS when it detects a syntax error or semantic error in the metadata 🛑.
The SRS MAY explicitly request a full metadata snapshot by sending an UPDATE request 🙋♀️. This request MUST contain a body with Content-Disposition type "recording-session" and MUST NOT contain an SDP body 🚫. The SRS MUST NOT request a full metadata snapshot in an UPDATE response or in any other SIP transaction 🙅♀️. The format of the content is "application/rs-metadata", and the body is an XML document whose format is defined in [RFC7865]. Figure 13 shows an example 👇:
UPDATE sip:2000@src.example.com SIP/2.0
Via: SIP/2.0/UDP srs.example.com;branch=z9hG4bKdf6b622b648d9
To: <sip:2000@example.com>;tag=35e195d2-947d-4585-946f-098392474
From: <sip:recorder@example.com>;tag=1234567890
Call-ID: d253c800-b0d1ea39-4a7dd-3f0e20a
CSeq: 1 UPDATE
Max-Forwards: 70
Require: siprec
Contact: <sip:recorder@srs.example.com>;+sip.srs
Accept: application/sdp, application/rs-metadata
Content-Disposition: recording-session
Content-Type: application/rs-metadata
Content-Length: [length]
<?xml version="1.0" encoding="UTF-8"?>
<requestsnapshot xmlns='urn:ietf:params:xml:ns:recording:1'>
<requestreason xml:lang="it">SRS internal error</requestreason>
</requestsnapshot>
Figure 13: Metadata RequestNote that UPDATE was chosen for the SRS to request a metadata snapshot, because it can be sent regardless of the state of the dialog 🌷. This was seen as better than requiring support for both UPDATE and re-INVITE messages for this operation 💫.
When the SRC receives a request for a metadata snapshot, it MUST immediately provide a full metadata snapshot in a separate INVITE or UPDATE transaction 🎯. Any subsequent partial updates won't depend on any metadata sent before this full metadata snapshot 🙌.
The metadata the SRS receives can contain ID elements used to cross-reference one element to another 🔗. An element containing the definition of an ID and an element containing a reference to that ID will often come from the same SRC 💕. It's also valid for those elements to come from different SRCs, for example when each endpoint in the same CS acts as an SRC to record the call and a common ID refers to the same CS 🤝. The SRS MUST NOT consider this an error 🥰.
10. Persistent Recording ✨
Okay bestie, persistent recording is a specific use case that addresses REQ-005 in [RFC6341], where an RS can be set up even when there's no CS 🦋. The SRC keeps recording media into an RS and sending it to the SRS even when there's no CS, for all the UAs that are part of persistent recording. Because the recorded streams are already allocated and the media keeps flowing to the SRS, the SRC doesn't have to build new recorded streams with a new SDP offer every time a new CS shows up. It also doesn't mess with the timing of the CS 💅. The only thing the SRC needs to update is the metadata when new CSs get created.
When there's no CS running on the devices with persistent recording, there's no recorded media to stream from the SRC to the SRS. In some environments that use a Network Address Translator (NAT), a minimum amount of flow activity is usually needed to keep the NAT binding alive for each port that's opened 🌈. Agents that support Interactive Connectivity Establishment (ICE) already solve this problem 🧚. For non-ICE agents, the SRC and SRS SHOULD follow the recommendations in [RFC6263] to maintain the NAT bindings, so they don't lose the bindings for the RTP/RTCP ports opened for the recorded streams 🫶.
11. IANA Considerations 🎀
11.1. Registration of Option Tags
This specification registers two option tags. The required information for this registration, as specified in [RFC3261], is as follows 💖.
11.1.1. "siprec" Option Tag
- Name:
siprec - Description: This option tag is for identifying that the SIP session is for the purpose of an RS. It's typically not used in a
Supportedheader. When it's present in aRequireheader in a request, it means the UA is either an SRC or SRS capable of handling an RS. 🌸
11.1.2. "record-aware" Option Tag
- Name:
record-aware - Description: This option tag indicates the UA's ability to receive recording indicators in media-level or session-level SDP. When it's present in a
Supportedheader, it means the UA can receive recording indicators in media-level or session-level SDP ✨.
11.2. Registration of Media Feature Tags
This document registers two new media feature tags in the SIP tree, following the process defined in [RFC2506] and [RFC3840] 🧁.
11.2.1. Feature Tag for the SRC
- Media feature tag name:
sip.src - ASN.1 Identifier:
1.3.6.1.8.4.27 - Summary of the media feature indicated by this tag: This feature tag says the UA is a Session Recording Client for the purpose of an RS.
- Values appropriate for use with this feature tag: boolean
- The feature tag is intended primarily for use in the following applications, protocols, services, or negotiation mechanisms: This feature tag is only useful for an RS.
- Examples of typical use: Routing the request to a Session Recording Server.
- Security Considerations: Security considerations for this media feature tag are discussed in Section 11.1 of RFC 3840.
11.2.2. Feature Tag for the SRS
- Media feature tag name:
sip.srs - ASN.1 Identifier:
1.3.6.1.8.4.28 - Summary of the media feature indicated by this tag: This feature tag says the UA is a Session Recording Server for the purpose of an RS.
- Values appropriate for use with this feature tag: boolean
- The feature tag is intended primarily for use in the following applications, protocols, services, or negotiation mechanisms: This feature tag is only useful for an RS.
- Examples of typical use: Routing the request to a Session Recording Client.
- Security Considerations: Security considerations for this media feature tag are discussed in Section 11.1 of RFC 3840.
11.3. New Content-Disposition Parameter Registrations
This document registers a new "disposition-type" value in the Content-Disposition header: recording-session 📜.
recording-session: The body describes either:- metadata about the RS, or
- the reason for the metadata snapshot request,
as determined by the MIME value indicated in the Content-Type.
11.4. SDP Attributes
This document registers the following new SDP attributes 🦋.
11.4.1. "record" SDP Attribute
- Contact names: Leon Portman,
leon.portman@nice.com; Henry Lum,henry.lum@genesyslab.com - Attribute name:
record - Long-form attribute name: Recording Indication
- Type of attribute: session level or media level
- Subject to charset: no
- Allowed attribute values:
on,off,paused
This attribute provides the recording indication for the session or media stream 🎀.
11.4.2. "recordpref" SDP Attribute
- Contact names: Leon Portman,
leon.portman@nice.com; Henry Lum,henry.lum@genesyslab.com - Attribute name:
recordpref - Long-form attribute name: Recording Preference
- Type of attribute: session level or media level
- Subject to charset: no
- Allowed attribute values:
on,off,pause,nopreference
This attribute provides the recording preference for the session or media stream 💜.
12. Security Considerations 🔐
This part is seriously important, bestie, so read it slowly 😭.
The RS is basically a standard SIP dialog [RFC3261], so it can reuse any of the existing SIP security mechanisms for securing the session signaling, the recorded media, and the metadata. The use cases and requirements document [RFC6341] outlines the general security considerations, and this document describes the specific security recommendations.
The SRC and SRS MUST support SIP with Transport Layer Security (TLS) version 1.2, SHOULD follow the best practices when using TLS as per [RFC7525], and MAY use Session Initiation Protocol Secure (SIPS) with TLS as per [RFC5630]. The RS MUST be at least as secure as the CS. This means using at least the same strength of cipher suite as the CS if the CS is secured. For example, if the CS uses SIPS for signaling and RTP/SAVP for media, then the RS may not use SIP or plain RTP unless other equivalent security measures are in effect, since doing so would mean an effective security downgrade 💀. Examples of other potentially equivalent security mechanisms include mutually authenticated TLS for the RS signaling channel or an appropriately protected network path for the RS media component.
12.1. Authentication and Authorization
At the transport level, the RS uses TLS authentication to validate the authenticity of the SRC and SRS. The SRC and SRS MUST implement TLS mutual authentication for establishing the RS. Whether the SRC/SRS chooses to use TLS mutual authentication is a deployment decision. In deployments where a UA acts as its own SRC, this requires that the UA have its own certificate as needed for TLS mutual authentication. In deployments where the SRC and the SRS are in the same administrative domain and have some other means of assuring authenticity, the SRC and SRS may choose not to authenticate each other, or to have the SRC authenticate the SRS only. In deployments where the SRS can be hosted on a different administrative domain, it is important to perform mutual authentication to ensure the authenticity of both the SRC and the SRS before transmitting any recorded media. The risk of not authenticating the SRS is that the recording may be sent to an entity other than the intended SRS, allowing a sensitive call recording to be received by an attacker 🚨. On the other hand, the risk of not authenticating the SRC is that an SRS will accept calls from an unknown SRC and allow potential forgery of call recordings 🙈.
There may be scenarios in which the signaling between the SRC and SRS is not direct, e.g., a SIP proxy exists between the SRC and the SRS. In such scenarios, each hop is subject to the TLS mutual authentication constraint, and transitive trust at each hop is utilized. Additionally, an SRC or SRS may use other existing SIP mechanisms available, including, but not limited to, Digest authentication [RFC3261], asserted identity [RFC3325], and connected identity [RFC4916] 🔑.
The SRS may have its own set of recording policies to authorize recording requests from the SRC. The use of recording policies is outside the scope of the Session Recording Protocol 🌈.
12.2. RTP Handling
In many scenarios, it will be critical for the media transported between the SRC and the SRS to be protected. Media encryption is an important element in the overall SIPREC solution. Therefore, the SRC and the SRS MUST support RTP/SAVP [RFC3711] and RTP/SAVPF [RFC5124]. RTP/SAVP and RTP/SAVPF provide media encryption, integrity protection, replay protection, and a limited form of source authentication. They do not contain or require a specific keying mechanism. At a minimum, the SRC and SRS MUST support the SDP security descriptions key negotiation mechanism [RFC4568]. For cases in which Datagram Transport Layer Security for Secure RTP (DTLS-SRTP) is used to encrypt a CS media stream, an SRC may use SRTP Encrypted Key Transport (EKT) [EKT-SRTP] in order to use SRTP-SDES in the RS without needing to re-encrypt the media ✨.
Note: When using EKT in this manner, it is possible for participants in the CS to send traffic that appears to be from other participants and have this forwarded by the SRC to the SRS within the RS. If this is a concern (e.g., the RS is intended for audit or compliance purposes), EKT is not an appropriate choice.
When RTP/SAVP or RTP/SAVPF is used, an SRC can choose to use the same keys or different keys in the RS than those used in the CS. Some SRCs are designed to simply replicate RTP packets from a CS media stream to the SRS. In that case, the SRC will use the same key in the RS as the key used in the CS. In this case, the SRC MUST secure the SDP containing the keying material in the RS with at least the same level of security as in the CS. The risk of lowering the level of security in the RS is that it will effectively become a downgrade attack on the CS, since the same key is used for both the CS and the RS 😱.
SRCs that decrypt an encrypted CS media stream and re-encrypt it when sending it to the SRS MUST use a different key than what is used for the CS media stream. This ensures that it is not possible for someone who has the key for the CS media stream to access recorded data they are not authorized to access. In order to maintain a comparable level of security, the key used in the RS SHOULD be of equivalent strength to, or greater strength than, that used in the CS 🔐.
12.3. Metadata
Metadata contains sensitive information, such as the address of record of the participants and other extension data placed by the SRC. It is essential to protect the content of the metadata in the RS 🫣. Since metadata is a content type transmitted in SIP signaling, metadata SHOULD be protected at the transport level by SIPS/TLS.
12.4. Storage and Playback
While storage and playback of the call recording are beyond the scope of this document, it is worthwhile to mention here that it is also important for the recording storage and playback to provide a level of security that is comparable to the CS. It would defeat the purpose of securing both the CS and the RS mentioned in the previous sections if the recording can be easily played back with a simple, unsecured HTTP interface without any form of authentication or authorization 🙅♀️.
13. References 📚
13.1. Normative References 📚
These are the must-reads the spec leans on, so here they are, bestie 💅
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997, <http://www.rfc-editor.org/info/rfc2119>.
[RFC2506] Holtman, K., Mutz, A., and T. Hardie, "Media Feature Tag Registration Procedure", BCP 31, RFC 2506, DOI 10.17487/RFC2506, March 1999, <http://www.rfc-editor.org/info/rfc2506>.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, DOI 10.17487/RFC3261, June 2002, <http://www.rfc-editor.org/info/rfc3261>.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session Description Protocol (SDP)", RFC 3264, DOI 10.17487/RFC3264, June 2002, <http://www.rfc-editor.org/info/rfc3264>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3840] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Indicating User Agent Capabilities in the Session Initiation Protocol (SIP)", RFC 3840, DOI 10.17487/RFC3840, August 2004, <http://www.rfc-editor.org/info/rfc3840>.
[RFC4574] Levin, O. and G. Camarillo, "The Session Description Protocol (SDP) Label Attribute", RFC 4574, DOI 10.17487/RFC4574, August 2006, <http://www.rfc-editor.org/info/rfc4574>.
[RFC5234] Crocker, D., Ed., and P. Overell, "Augmented BNF for Syntax Specifications: ABNF", STD 68, RFC 5234, DOI 10.17487/RFC5234, January 2008, <http://www.rfc-editor.org/info/rfc5234>.
[RFC7245] Hutton, A., Ed., Portman, L., Ed., Jain, R., and K. Rehor, "An Architecture for Media Recording Using the Session Initiation Protocol", RFC 7245, DOI 10.17487/RFC7245, May 2014, <http://www.rfc-editor.org/info/rfc7245>.
[RFC7865] Ravindranath, R., Ravindran, P., and P. Kyzivat, "Session Initiation Protocol (SIP) Recording Metadata", RFC 7865, DOI 10.17487/RFC7865, May 2016, <http://www.rfc-editor.org/info/rfc7865>.
13.2. Informative References 🌸
Extra reading for when you want the full picture 🧚
[EKT-SRTP] Mattsson, J., Ed., McGrew, D., Wing, D., and F. Andreasen, "Encrypted Key Transport for Secure RTP", Work in Progress, draft-ietf-avtcore-srtp-ekt-03, October 2014.
[RFC2804] IAB and IESG, "IETF Policy on Wiretapping", RFC 2804, DOI 10.17487/RFC2804, May 2000, <http://www.rfc-editor.org/info/rfc2804>.
[RFC3311] Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE Method", RFC 3311, DOI 10.17487/RFC3311, October 2002, <http://www.rfc-editor.org/info/rfc3311>.
[RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks", RFC 3325, DOI 10.17487/RFC3325, November 2002, <http://www.rfc-editor.org/info/rfc3325>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video Conferences with Minimal Control", STD 65, RFC 3551, DOI 10.17487/RFC3551, July 2003, <http://www.rfc-editor.org/info/rfc3551>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, DOI 10.17487/RFC3711, March 2004, <http://www.rfc-editor.org/info/rfc3711>.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session Description Protocol (SDP) Security Descriptions for Media Streams", RFC 4568, DOI 10.17487/RFC4568, July 2006, <http://www.rfc-editor.org/info/rfc4568>.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, DOI 10.17487/RFC4585, July 2006, <http://www.rfc-editor.org/info/rfc4585>.
[RFC4916] Elwell, J., "Connected Identity in the Session Initiation Protocol (SIP)", RFC 4916, DOI 10.17487/RFC4916, June 2007, <http://www.rfc-editor.org/info/rfc4916>.
[RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)", BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007, <http://www.rfc-editor.org/info/rfc4961>.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, "Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104, February 2008, <http://www.rfc-editor.org/info/rfc5104>.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February 2008, <http://www.rfc-editor.org/info/rfc5124>.
[RFC5168] Levin, O., Even, R., and P. Hagendorf, "XML Schema for Media Control", RFC 5168, DOI 10.17487/RFC5168, March 2008, <http://www.rfc-editor.org/info/rfc5168>.
[RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session Initiation Protocol (SIP)", RFC 5630, DOI 10.17487/RFC5630, October 2009, <http://www.rfc-editor.org/info/rfc5630>.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and Control Packets on a Single Port", RFC 5761, DOI 10.17487/RFC5761, April 2010, <http://www.rfc-editor.org/info/rfc5761>.
[RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for Keeping Alive the NAT Mappings Associated with RTP / RTP Control Protocol (RTCP) Flows", RFC 6263, DOI 10.17487/RFC6263, June 2011, <http://www.rfc-editor.org/info/rfc6263>.
[RFC6341] Rehor, K., Ed., Portman, L., Ed., Hutton, A., and R. Jain, "Use Cases and Requirements for SIP-Based Media Recording (SIPREC)", RFC 6341, DOI 10.17487/RFC6341, August 2011, <http://www.rfc-editor.org/info/rfc6341>.
[RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla, "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022, September 2013, <http://www.rfc-editor.org/info/rfc7022>.
[RFC7525] Sheffer, Y., Holz, R., and P. Saint-Andre, "Recommendations for Secure Use of Transport Layer Security (TLS) and Datagram Transport Layer Security (DTLS)", BCP 195, RFC 7525, DOI 10.17487/RFC7525, May 2015, <http://www.rfc-editor.org/info/rfc7525>.
Acknowledgements 💖
Huge thank-yous to John Elwell, Paul Kyzivat, Partharsarathi R, Ram Mohan R, Hadriel Kaplan, Adam Roach, Miguel Garcia, Thomas Stach, Muthu Perumal, Dan Wing, and Magnus Westerlund for their valuable comments and inputs to this document 🫶✨
Authors' Addresses 🎀
Leon Portman
NICE Systems
22 Zarhin Street
P.O. Box 690
Ra'anana 4310602
Israel
Email: leon.portman@gmail.com
Henry Lum (editor)
Genesys
1380 Rodick Road, Suite 201
Markham, Ontario L3R4G5
Canada
Email: henry.lum@genesyslab.com
Charles Eckel
Cisco
170 West Tasman Drive
San Jose, CA 95134
United States
Email: eckelcu@cisco.com
Alan Johnston
Illinois Institute of Technology
Bellevue, WA
United States
Email: alan.b.johnston@gmail.com
Andrew Hutton
Unify
Brickhill Street
Milton Keynes MK15 0DJ
United Kingdom
Email: andrew.hutton@unify.com📦 where this page comes from
- ietf: rfc7866 2016-05
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