Internet Engineering Task Force (IETF)
Request for Comments: 7657
Category: Informational
ISSN: 2070-1721
D. Black, Ed.
P. Jones
November 2015

Differentiated Services (Diffserv) and Real-Time Communication


This memo describes the interaction between Differentiated Services (Diffserv) network quality-of-service (QoS) functionality and real- time network communication, including communication based on the Real-time Transport Protocol (RTP). Diffserv is based on network nodes applying different forwarding treatments to packets whose IP headers are marked with different Diffserv Codepoints (DSCPs). WebRTC applications, as well as some conferencing applications, have begun using the Session Description Protocol (SDP) bundle negotiation mechanism to send multiple traffic streams with different QoS requirements using the same network 5-tuple. The results of using multiple DSCPs to obtain different QoS treatments within a single network 5-tuple have transport protocol interactions, particularly with congestion control functionality (e.g., reordering). In addition, DSCP markings may be changed or removed between the traffic source and destination. This memo covers the implications of these Diffserv aspects for real-time network communication, including WebRTC.

Status of This Memo

This document is not an Internet Standards Track specification; it is published for informational purposes.

This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Not all documents approved by the IESG are a candidate for any level of Internet Standard; see Section 2 of RFC 5741.

Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at

Copyright Notice

Copyright © 2015 IETF Trust and the persons identified as the document authors. All rights reserved.

This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents ( in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Real-Time Communications  . . . . . . . . . . . . . . . . . .   3
     2.1.  RTP Background  . . . . . . . . . . . . . . . . . . . . .   4
     2.2.  RTP Multiplexing  . . . . . . . . . . . . . . . . . . . .   6
   3.  Differentiated Services (Diffserv)  . . . . . . . . . . . . .   7
     3.1.  Diffserv Per-Hop Behaviors (PHBs) . . . . . . . . . . . .  10
     3.2.  Traffic Classifiers and DSCP Remarking  . . . . . . . . .  10
   4.  Examples  . . . . . . . . . . . . . . . . . . . . . . . . . .  12
   5.  Diffserv Interactions . . . . . . . . . . . . . . . . . . . .  13
     5.1.  Diffserv, Reordering, and Transport Protocols . . . . . .  13
     5.2.  Diffserv, Reordering, and Real-Time Communication . . . .  15
     5.3.  Drop Precedence and Transport Protocols . . . . . . . . .  16
     5.4.  Diffserv and RTCP . . . . . . . . . . . . . . . . . . . .  17
   6.  Guidelines  . . . . . . . . . . . . . . . . . . . . . . . . .  18
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  19
   8.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  20
     8.1.  Normative References  . . . . . . . . . . . . . . . . . .  20
     8.2.  Informative References  . . . . . . . . . . . . . . . . .  22
   Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . .  26
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  26

1. Introduction

This memo describes the interactions between Differentiated Services (Diffserv) network quality-of-service (QoS) functionality [RFC2475] and real-time network communication, including communication based on the Real-time Transport Protocol (RTP) [RFC3550]. Diffserv is based on network nodes applying different forwarding treatments to packets whose IP headers are marked with different Diffserv Codepoints (DSCPs) [RFC2474]. In the past, distinct RTP streams have been sent over different transport-level flows, sometimes multiplexed with the RTP Control Protocol (RTCP). WebRTC applications, as well as some conferencing applications, are now using the Session Description Protocol (SDP) [RFC4566] bundle negotiation mechanism [SDP-BUNDLE] to send multiple traffic streams with different QoS requirements using the same network 5-tuple. The results of using multiple DSCPs to obtain different QoS treatments within a single network 5-tuple have transport protocol interactions, particularly with congestion control functionality (e.g., reordering). In addition, DSCP markings may be changed or removed between the traffic source and destination. This memo covers the implications of these Diffserv aspects for real-time network communication, including WebRTC traffic [WEBRTC-OVERVIEW].

The memo is organized as follows. Background is provided in Section 2 on real-time communications and Section 3 on Differentiated Services. Section 4 describes some examples of Diffserv usage with real-time communications. Section 5 explains how use of Diffserv features interacts with both transport and real-time communications protocols and Section 6 provides guidance on Diffserv feature usage to control undesired interactions. Security considerations are discussed in Section 7.

2. Real-Time Communications

Real-time communications enables communication in real time over an IP network using voice, video, text, content sharing, etc. It is possible to use more than one of these modes concurrently to provide a rich communication experience.

A simple example of real-time communications is a voice call placed over the Internet where an audio stream is transmitted in each direction between two users. A more complex example is an immersive videoconferencing system that has multiple video screens, multiple cameras, multiple microphones, and some means of sharing content. For such complex systems, there may be multiple media and non-media streams transmitted via a single IP address and port or via multiple IP addresses and ports.

2.1. RTP Background

The most common protocol used for real-time media is RTP [RFC3550]. RTP defines a common encapsulation format and handling rules for real-time data transmitted over the Internet. Unfortunately, RTP terminology usage has been inconsistent. For example, RFC 7656 [RFC7656] on RTP terminology observes that:

RTP [RFC3550] uses media stream, audio stream, video stream, and a stream of (RTP) packets interchangeably, which are all RTP streams.

Terminology in this memo is based on that RTP terminology document with the following terms being of particular importance (see that terminology document for full definitions):

   Source Stream:  A reference clock synchronized, time progressing,
      digital media stream.
   RTP Stream:  A stream of RTP packets containing media data, which may
      be source data or redundant data.  The RTP stream is identified by
      an RTP synchronization source (SSRC) belonging to a particular RTP
      session.  An RTP stream may be a secured RTP stream when RTP-based
      security is used.

In addition, this memo follows [RFC3550] in using the term "SSRC" to designate both the identifier of an RTP stream and the entity that sends that RTP stream.

Media encoding and packetization of a source stream results in a source RTP stream plus zero or more redundancy RTP streams that provide resilience against loss of packets from the source RTP stream [RFC7656]. Redundancy information may also be carried in the same RTP stream as the encoded source stream, e.g., see Section 7.2 of [RFC5109]. With most applications, a single media type (e.g., audio) is transmitted within a single RTP session. However, it is possible to transmit multiple, distinct source streams over the same RTP session as one or more individual RTP streams. This is referred to as RTP multiplexing. In addition, an RTP stream may contain multiple source streams, e.g., components or programs in an MPEG Transport Stream [H.221].

The number of source streams and RTP streams in an overall real-time interaction can be surprisingly large. In addition to a voice source stream and a video source stream, there could be separate source streams for each of the cameras or microphones on a videoconferencing system. As noted above, there might also be separate redundancy RTP streams that provide protection to a source RTP stream, using techniques such as forward error correction. Another example is simulcast transmission, where a video source stream can be transmitted as high resolution and low resolution RTP streams at the same time. In this case, a media processing function might choose to send one or both RTP streams onward to a receiver based on bandwidth availability or who the active speaker is in a multipoint conference. Lastly, a transmitter might send the same media content concurrently as two RTP streams using different encodings (e.g., video encoded as VP8 [RFC6386] in parallel with H.264 [H.264]) to allow a media processing function to select a media encoding that best matches the capabilities of the receiver.

For the WebRTC protocol suite [WEBRTC-TRANSPORTS], an individual source stream is a MediaStreamTrack, and a MediaStream contains one or more MediaStreamTracks [W3C.WD-mediacapture-streams-20130903]. A MediaStreamTrack is transmitted as a source RTP stream plus zero or more redundant RTP streams, so a MediaStream that consists of one MediaStreamTrack is transmitted as a single source RTP stream plus zero or more redundant RTP streams. For more information on use of RTP in WebRTC, see [RTP-USAGE].

RTP is usually carried over a datagram protocol, such as UDP [RFC768], UDP-Lite [RFC3828], or the Datagram Congestion Control Protocol (DCCP) [RFC4340]; UDP is most commonly used, but a non- datagram protocol (e.g., TCP [RFC793]) may also be used. Transport protocols other than UDP or UDP-Lite may also be used to transmit real-time data or near-real-time data. For example, the Stream Control Transmission Protocol (SCTP) [RFC4960] can be utilized to carry application-sharing or whiteboarding information as part of an overall interaction that includes real-time media. These additional transport protocols can be multiplexed with an RTP session via UDP encapsulation, thereby using a single pair of UDP ports.

The WebRTC protocol suite encompasses a number of forms of multiplexing:

  1. Individual source streams are carried in one or more individual RTP streams. These RTP streams can be multiplexed onto a single transport-layer flow or sent as separate transport-layer flows. This memo only considers the case where the RTP streams are to be multiplexed onto a single transport-layer flow, forming a single RTP session as described in [RFC3550];
   2.  RTCP (see [RFC3550]) may be multiplexed onto the same transport-
       layer flow as the RTP streams with which it is associated, as
       described in [RFC5761], or it may be sent on a separate
       transport-layer flow;
  1. An RTP session could be multiplexed with a single SCTP association over Datagram Transport Layer Security (DTLS) and with both Session Traversal Utilities for NAT (STUN) [RFC5389] and TURN [RFC5766] traffic into a single transport-layer flow as described in [RFC5764] with the updates in [SRTP-DTLS]. The STUN [RFC5389] and Traversal Using Relays around NAT (TURN) [RFC5766] protocols provide NAT/FW (Network Address Translator / Firewall) traversal and port mapping.

The resulting transport-layer flow is identified by a network 5-tuple, i.e., a combination of two IP addresses (source and destination), two ports (source and destination), and the transport protocol used (e.g., UDP). SDP bundle negotiation restrictions [SDP-BUNDLE] limit WebRTC to using at most a single DTLS session per network 5-tuple. In contrast to WebRTC use of a single SCTP association with DTLS, multiple SCTP associations can be directly multiplexed over a single UDP 5-tuple as specified in [RFC6951].

The STUN and TURN protocols were originally designed to use UDP as a transport; however, TURN has been extended to use TCP as a transport for situations in which UDP does not work [RFC6062]. When TURN selects use of TCP, the entire real-time communications session is carried over a single TCP connection (i.e., 5-tuple).

For IPv6, addition of the flow label [RFC6437] to network 5-tuples results in network 6-tuples (or 7-tuples for bidirectional flows), but in practice, use of a flow label is unlikely to result in a finer-grain traffic subset than the corresponding network 5-tuple (e.g., the flow label is likely to represent the combination of two ports with use of the UDP protocol). For that reason, discussion in this document focuses on UDP 5-tuples.

2.2. RTP Multiplexing

Section 2.1 explains how source streams can be multiplexed in a single RTP session, which can in turn be multiplexed over UDP with packets generated by other transport protocols. This section provides background on why this level of multiplexing is desirable. The rationale in this section applies both to multiplexing of source streams in a single RTP session and multiplexing of an RTP session with traffic from other transport protocols via UDP encapsulation.

Multiplexing reduces the number of ports utilized for real-time and related communication in an overall interaction. While a single endpoint might have plenty of ports available for communication, this traffic often traverses points in the network that are constrained on the number of available ports or whose performance degrades as the number of ports in use increases. A good example is a NAT/FW device sitting at the network edge. As the number of simultaneous protocol sessions increases, so does the burden placed on these devices to provide port mapping.

Another reason for multiplexing is to help reduce the time required to establish bidirectional communication. Since any two communicating users might be situated behind different NAT/FW devices, it is necessary to employ techniques like STUN and TURN along with Interactive Connectivity Establishment (ICE) [RFC5245] to get traffic to flow between the two devices [WEBRTC-TRANSPORTS]. Performing the tasks required by these protocols takes time, especially when multiple protocol sessions are involved. While tasks for different sessions can be performed in parallel, it is nonetheless necessary for applications to wait for all sessions to be opened before communication between two users can begin. Reducing the number of STUN/ICE/TURN steps reduces the likelihood of loss of a packet for one of these protocols; any such loss adds delay to setting up a communication session. Further, reducing the number of STUN/ICE/TURN tasks places a lower burden on the STUN and TURN servers.

Multiplexing may reduce the complexity and resulting load on an endpoint. A single instance of STUN/ICE/TURN is simpler to execute and manage than multiple instances STUN/ICE/TURN operations happening in parallel, as the latter require synchronization and create more complex failure situations that have to be cleaned up by additional code.

3. Differentiated Services (Diffserv)

The Diffserv architecture [RFC2475][RFC4594] is intended to enable scalable service discrimination in the Internet without requiring each node in the network to store per-flow state and participate in per-flow signaling. The services may be end to end or within a network; they include both those that can satisfy quantitative performance requirements (e.g., peak bandwidth) and those based on relative performance (e.g., "class" differentiation). Services can be constructed by a combination of well-defined building blocks deployed in network nodes that:

  • classify traffic and set bits in an IP header field at network boundaries or hosts,
  • use those bits to determine how packets are forwarded by the nodes inside the network, and
  • condition the marked packets at network boundaries in accordance with the requirements or rules of each service.

Traffic conditioning may include changing the DSCP in a packet (remarking it), delaying the packet (as a consequence of traffic shaping), or dropping the packet (as a consequence of traffic policing).

A network node that supports Diffserv includes a classifier that selects packets based on the value of the DS field in IP headers (the Diffserv codepoint or DSCP), along with buffer management and packet scheduling mechanisms capable of delivering the specific packet forwarding treatment indicated by the DS field value. Setting of the DS field and fine-grain conditioning of marked packets need only be performed at network boundaries; internal network nodes operate on traffic aggregates that share a DS field value, or in some cases, a small set of related values.

The Diffserv architecture [RFC2475] maintains distinctions among:

  • the QoS service provided to a traffic aggregate,
  • the conditioning functions and per-hop behaviors (PHBs) used to realize services,
  • the DSCP in the IP header used to mark packets to select a per-hop behavior, and
  • the particular implementation mechanisms that realize a per-hop behavior.

This memo focuses on PHBs and the usage of DSCPs to obtain those behaviors. In a network node's forwarding path, the DSCP is used to map a packet to a particular forwarding treatment, or to a per-hop behavior (PHB) that specifies the forwarding treatment.

The specification of a PHB describes the externally observable forwarding behavior of a network node for network traffic marked with a DSCP that selects that PHB. In this context, "forwarding behavior" is a general concept - for example, if only one DSCP is used for all traffic on a link, the observable forwarding behavior (e.g., loss, delay, jitter) will often depend only on the loading of the link. To obtain useful behavioral differentiation, multiple traffic subsets are marked with different DSCPs for different PHBs for which node resources such as buffer space and bandwidth are allocated. PHBs provide the framework for a Diffserv network node to allocate resources to traffic subsets, with network-scope Differentiated Services constructed on top of this basic hop-by-hop resource allocation mechanism.

The codepoints (DSCPs) may be chosen from a small set of fixed values (the class selector codepoints), from a set of recommended values defined in PHB specifications, or from values that have purely local meanings to a specific network that supports Diffserv; in general, packets may be forwarded across multiple such networks between source and destination.

The mandatory DSCPs are the class selector codepoints as specified in [RFC2474]. The class selector codepoints (CS0-CS7) extend the deprecated concept of IP Precedence in the IPv4 header; three bits are added, so that the class selector DSCPs are of the form 'xxx000'. The all-zero DSCP ('000000' or CS0) is always assigned to a Default PHB that provides best-effort forwarding behavior, and the remaining class selector codepoints are intended to provide relatively better per-hop-forwarding behavior in increasing numerical order, but:

  • A network endpoint cannot rely upon different class selector codepoints providing Differentiated Services via assignment to different PHBs, as adjacent class selector codepoints may use the same pool of resources on each network node in some networks. This generalizes to ranges of class selector codepoints, but with limits -- for example, CS6 and CS7 are often used for network control (e.g., routing) traffic [RFC4594] and hence are likely to provide better forwarding behavior under network load to prioritize network recovery from disruptions. There is no effective way for a network endpoint to determine which PHBs are selected by the class selector codepoints on a specific network, let alone end to end.
  • CS1 ('001000') was subsequently designated as the recommended codepoint for the Lower Effort (LE) PHB [RFC3662]. An LE service forwards traffic with "lower" priority than best effort and can be "starved" by best-effort and other "higher" priority traffic. Not all networks offer an LE service, hence traffic marked with the CS1 DSCP may not receive lower effort forwarding; such traffic may be forwarded with a different PHB (e.g., the Default PHB), remarked to another DSCP (e.g., CS0) and forwarded accordingly, or dropped. A network endpoint cannot rely upon the presence of an LE service that is selected by the CS1 DSCP on a specific network, let alone end to end. Packets marked with the CS1 DSCP may be forwarded with best-effort service or another "higher" priority service; see [RFC2474]. See [RFC3662] for further discussion of the LE PHB and service.

3.1. Diffserv Per-Hop Behaviors (PHBs)

Although Differentiated Services is a general architecture that may be used to implement a variety of services, three fundamental forwarding behaviors (PHBs) have been defined and characterized for general use. These are:

   1.  Default Forwarding (DF) for elastic traffic [RFC2474].  The
       Default PHB is always selected by the all-zero DSCP and provides
       best-effort forwarding.
   2.  Assured Forwarding (AF) [RFC2597] to provide Differentiated
       Service to elastic traffic.  Each instance of the AF behavior
       consists of three PHBs that differ only in drop precedence, e.g.,
       AF11, AF12, and AF13; such a set of three AF PHBs is referred to
       as an AF class, e.g., AF1x.  There are four defined AF classes,
       AF1x through AF4x, with higher numbered classes intended to
       receive better forwarding treatment than lower numbered classes.
       Use of multiple PHBs from a single AF class (e.g., AF1x) does not
       enable network traffic reordering within a single network
       5-tuple, although such reordering may occur for other transient
       reasons (e.g., routing changes or ECMP rebalancing).
   3.  Expedited Forwarding (EF) [RFC3246] intended for inelastic
       traffic.  Beyond the basic EF PHB, the VOICE-ADMIT PHB [RFC5865]
       is an admission-controlled variant of the EF PHB.  Both of these
       PHBs are based on preconfigured limited forwarding capacity;
       traffic in excess of that capacity is expected to be dropped.

3.2. Traffic Classifiers and DSCP Remarking

DSCP markings are not end to end in general. Each network can make its own decisions about what PHBs to use and which DSCP maps to each PHB. While every PHB specification includes a recommended DSCP, and RFC 4594 [RFC4594] recommends their end-to-end usage, there is no requirement that every network support any PHBs (aside from the Default PHB for best-effort forwarding) or use any specific DSCPs, with the exception of the support requirements for the class selector codepoints (see RFC 2474 [RFC2474]). When Diffserv is used, the edge or boundary nodes of a network are responsible for ensuring that all traffic entering that network conforms to that network's policies for DSCP and PHB usage, and such nodes may change DSCP markings on traffic to achieve that result. As a result, DSCP remarking is possible at any network boundary, including the first network node that traffic sent by a host encounters. Remarking is also possible within a network, e.g., for traffic shaping.

DSCP remarking is part of traffic conditioning; the traffic conditioning functionality applied to packets at a network node is determined by a traffic classifier [RFC2475]. Edge nodes of a Diffserv network classify traffic based on selected packet header fields; typical implementations do not look beyond the traffic's network 5-tuple in the IP and transport protocol headers (e.g., for SCTP or RTP encapsulated in UDP, header-based classification is unlikely to look beyond the outer UDP header). As a result, when multiple DSCPs are used for traffic that shares a network 5-tuple, remarking at a network boundary may result in all of the traffic being forwarded with a single DSCP, thereby removing any differentiation within the network 5-tuple downstream of the remarking location. Network nodes within a Diffserv network generally classify traffic based solely on DSCPs, but may perform finer-grain traffic conditioning similar to that performed by edge nodes.

So, for two arbitrary network endpoints, there can be no assurance that the DSCP set at the source endpoint will be preserved and presented at the destination endpoint. Rather, it is quite likely that the DSCP will be set to zero (e.g., at the boundary of a network operator that distrusts or does not use the DSCP field) or to a value deemed suitable by an ingress classifier for whatever network 5-tuple it carries.

In addition, remarking may remove application-level distinctions in forwarding behavior - e.g., if multiple PHBs within an AF class are used to distinguish different types of frames within a video RTP stream, token-bucket-based remarkers operating in color-blind mode (see [RFC2697] and [RFC2698] for examples) may remark solely based on flow rate and burst behavior, removing the drop precedence distinctions specified by the source.

Backbone and other carrier networks may employ a small number of DSCPs (e.g., less than half a dozen) to manage a small number of traffic aggregates; hosts that use a larger number of DSCPs can expect to find that much of their intended differentiation is removed by such networks. Better results may be achieved when DSCPs are used to spread traffic among a smaller number of Diffserv-based traffic subsets or aggregates; see [DIFFSERV-INTERCON] for one proposal. This is of particular importance for MPLS-based networks due to the limited size of the Traffic Class (TC) field in an MPLS label [RFC5462] that is used to carry Diffserv information and the use of that TC field for other purposes, e.g., Explicit Congestion Notification (ECN) [RFC5129]. For further discussion on use of Diffserv with MPLS, see [RFC3270] and [RFC5127].

4. Examples

For real-time communications, one might want to mark the audio packets using EF and the video packets as AF41. However, a video conference receiving the audio packets significantly ahead of the video is not useful because lip sync is necessary between audio and video. It may still be desirable to send audio with a PHB that provides better service, because more reliable arrival of audio helps assure smooth audio rendering, which is often more important than fully faithful video rendering. There are also limits, as some devices have difficulties in synchronizing voice and video when packets that need to be rendered together arrive at significantly different times. It makes more sense to use different PHBs when the audio and video source streams do not share a strict timing relationship. For example, video content may be shared within a video conference via playback, perhaps of an unedited video clip that is intended to become part of a television advertisement. Such content sharing video does not need precise synchronization with video conference audio, and could use a different PHB, as content sharing video is more tolerant to jitter, loss, and delay.

Within a layered video RTP stream, ordering of frame communication is preferred, but importance of frame types varies, making use of PHBs with different drop precedences appropriate. For example, I-frames that contain an entire image are usually more important than P-frames that contain only changes from the previous image because loss of a P-frame (or part thereof) can be recovered (at the latest) via the next I-frame, whereas loss of an I-frame (or part thereof) may cause rendering problems for all of the P-frames that depend on the missing I-frame. For this reason, it is appropriate to mark I-frame packets with a PHB that has lower drop precedence than the PHB used for P-frames, as long as the PHBs preserve ordering among frames (e.g., are in a single AF class) - AF41 for I-frames and AF43 for P-frames is one possibility. Additional spatial and temporal layers beyond the base video layer could also be marked with higher drop precedence than the base video layer, as their loss reduces video quality, but does not disrupt video rendering.

Additional RTP streams in a real-time communication interaction could be marked with CS0 and carried as best-effort traffic. One example is real-time text transmitted as specified in RFC 4103 [RFC4103]. Best-effort forwarding suffices because such real-time text has loose timing requirements; RFC 4103 recommends sending text in chunks every 300 ms. Such text is technically real-time, but does not need a PHB promising better service than best effort, in contrast to audio or video.

A WebRTC application may use one or more RTP streams, as discussed above. In addition, it may use an SCTP-based data channel [DATA-CHAN] whose QoS treatment depends on the nature of the application. For example, best-effort treatment of data channels is likely to suffice for messaging, shared white board, and guided browsing applications, whereas latency-sensitive games might desire better QoS for their data channels.

5. Diffserv Interactions

5.1. Diffserv, Reordering, and Transport Protocols

Transport protocols provide data communication behaviors beyond those possible at the IP layer. An important example is that TCP [RFC793] provides reliable in-order delivery of data with congestion control. SCTP [RFC4960] provides additional properties such as preservation of message boundaries, and the ability to avoid head-of-line blocking that may occur with TCP.

In contrast, UDP [RFC768] is a basic unreliable datagram protocol that provides port-based multiplexing and demultiplexing on top of IP. Two other unreliable datagram protocols are UDP-Lite [RFC3828], a variant of UDP that may deliver partially corrupt payloads when errors occur, and DCCP [RFC4340], which provides a range of congestion control modes for its unreliable datagram service.

Transport protocols that provide reliable delivery (e.g., TCP, SCTP) are sensitive to network reordering of traffic. When a protocol that provides reliable delivery receives a packet other than the next expected packet, the protocol usually assumes that the expected packet has been lost and updates the peer, which often causes a retransmission. In addition, congestion control functionality in transport protocols (including DCCP) usually infers congestion when packets are lost. This creates additional sensitivity to significant network packet reordering, as such reordering may be (mis)interpreted as loss of the out-of-order packets, causing a congestion control response.

This sensitivity to reordering remains even when ECN [RFC3168] is in use, as ECN receivers are required to treat missing packets as potential indications of congestion, because:

  • Severe congestion may cause ECN-capable network nodes to drop packets, and
  • ECN traffic may be forwarded by network nodes that do not support ECN and hence drop packets to indicate congestion.

Congestion control is an important aspect of the Internet architecture; see [RFC2914] for further discussion.

In general, marking packets with different DSCPs results in different PHBs being applied at nodes in the network, making reordering very likely due to use of different pools of forwarding resources for each PHB. This should not be done within a single network 5-tuple for current transport protocols, with the important exceptions of UDP and UDP-Lite.

When PHBs that enable reordering are mixed within a single network 5-tuple, the effect is to mix QoS-based traffic classes within the scope of a single transport protocol connection or association. As these QoS-based traffic classes receive different network QoS treatments, they use different pools of network resources and hence may exhibit different levels of congestion. The result for congestion-controlled protocols is that a separate instance of congestion control functionality is needed per QoS-based traffic class. Current transport protocols support only a single instance of congestion control functionality for an entire connection or association; extending that support to multiple instances would add significant protocol complexity. Traffic in different QoS-based classes may use different paths through the network; this complicates path integrity checking in connection- or association-based protocols, as those paths may fail independently.

The primary example where usage of multiple PHBs does not enable reordering within a single network 5-tuple is use of PHBs from a single AF class (e.g., AF1x). Traffic reordering within the scope of a network 5-tuple that uses a single PHB or AF class may occur for other transient reasons (e.g., routing changes or ECMP rebalancing).

Reordering also affects other forms of congestion control, such as techniques for RTP congestion control that were under development when this memo was published; see [RMCAT-CC] for requirements. These techniques prefer use of a common (coupled) congestion controller for RTP streams between the same endpoints to reduce packet loss and delay by reducing competition for resources at any shared bottleneck.

Shared bottlenecks can be detected via techniques such as correlation of one-way delay measurements across RTP streams. An alternate approach is to assume that the set of packets on a single network 5-tuple marked with DSCPs that do not enable reordering will utilize a common network path and common forwarding resources at each network node. Under that assumption, any bottleneck encountered by such packets is shared among all of them, making it safe to use a common (coupled) congestion controller (see [COUPLED-CC]). This is not a safe assumption when the packets involved are marked with DSCP values that enable reordering because a bottleneck may not be shared among all such packets (e.g., when the DSCP values result in use of different queues at a network node, but only one queue is a bottleneck).

UDP and UDP-Lite are not sensitive to reordering in the network, because they do not provide reliable delivery or congestion control. On the other hand, when used to encapsulate other protocols (e.g., as UDP is used by WebRTC; see Section 2.1), the reordering considerations for the encapsulated protocols apply. For the specific usage of UDP by WebRTC, every encapsulated protocol (i.e., RTP, SCTP, and TCP) is sensitive to reordering as further discussed in this memo. In addition, [RFC5405] provides general guidelines for use of UDP (and UDP-Lite); the congestion control guidelines in that document apply to protocols encapsulated in UDP (or UDP-Lite).

5.2. Diffserv, Reordering, and Real-Time Communication

Real-time communications are also sensitive to network reordering of packets. Such reordering may lead to unneeded retransmission and spurious retransmission control signals (such as NACK) in reliable delivery protocols (see Section 5.1). The degree of sensitivity depends on protocol or stream timers, in contrast to reliable delivery protocols that usually react to all reordering.

Receiver jitter buffers have important roles in the effect of reordering on real-time communications:

  • Minor packet reordering that is contained within a jitter buffer usually has no effect on rendering of the received RTP stream because packets that arrive out of order are retrieved in order from the jitter buffer for rendering.
  • Packet reordering that exceeds the capacity of a jitter buffer can cause user-perceptible quality problems (e.g., glitches, noise) for delay-sensitive communication, such as interactive conversations for which small jitter buffers are necessary to preserve human perceptions of real-time interaction. Interactive real-time communication implementations often discard data that is sufficiently late so that it cannot be rendered in source stream order, making retransmission counterproductive. For this reason, implementations of interactive real-time communication often do not use retransmission.
  • In contrast, replay of recorded media can tolerate significantly longer delays than interactive conversations, so replay is likely to use larger jitter buffers than interactive conversations. These larger jitter buffers increase the tolerance of replay to reordering by comparison to interactive conversations. The size of the jitter buffer imposes an upper bound on replay tolerance to reordering but does enable retransmission to be used when the jitter buffer is significantly larger than the amount of data that can be expected to arrive during the round-trip latency for retransmission.

Network packet reordering has no effective upper bound and can exceed the size of any reasonable jitter buffer. In practice, the size of jitter buffers for replay is limited by external factors such as the amount of time that a human is willing to wait for replay to start.

5.3. Drop Precedence and Transport Protocols

Packets within the same network 5-tuple that use PHBs within a single AF class can be expected to draw upon the same forwarding resources on network nodes (e.g., use the same router queue), and hence use of multiple drop precedences within an AF class is not expected to cause latency variation. When PHBs within a single AF class are mixed within a flow, the resulting overall likelihood that packets will be dropped from that flow is a mix of the drop likelihoods of the PHBs involved.

There are situations in which drop precedences should not be mixed. A simple example is that there is little value in mixing drop precedences within a TCP connection, because TCP's ordered delivery behavior results in any drop requiring the receiver to wait for the dropped packet to be retransmitted. Any resulting delay depends on the RTT and not the packet that was dropped. Hence a single DSCP should be used for all packets in a TCP connection.

As a consequence, when TCP is selected for NAT/FW traversal (e.g., by TURN), a single DSCP should be used for all traffic on that TCP connection. An additional reason for this recommendation is that packetization for STUN/ICE/TURN occurs before passing the resulting packets to TCP; TCP resegmentation may result in a different packetization on the wire, breaking any association between DSCPs and specific data to which they are intended to apply.

SCTP [RFC4960] differs from TCP in a number of ways, including the ability to deliver messages in an order that differs from the order in which they were sent and support for unreliable streams. However, SCTP performs congestion control and retransmission across the entire association, and not on a per-stream basis. Although there may be advantages to using multiple drop precedence across SCTP streams or within an SCTP stream that does not use reliable ordered delivery, there is no practical operational experience in doing so (e.g., the SCTP sockets API [RFC6458] does not support use of more than one DSCP for an SCTP association). As a consequence, the impacts on SCTP protocol and implementation behavior are unknown and difficult to predict. Hence a single DSCP should be used for all packets in an SCTP association, independent of the number or nature of streams in that association. Similar reasoning applies to a DCCP connection; a single DSCP should be used because the scope of congestion control is the connection and there is no operational experience with using more than one DSCP. This recommendation may be revised in the future if experiments, analysis, and operational experience provide compelling reasons to change it.

Guidance on transport protocol design and implementation to provide support for use of multiple PHBs and DSCPs in a transport protocol connection (e.g., DCCP) or transport protocol association (e.g., SCTP) is out of scope for this memo.

5.4. Diffserv and RTCP

RTCP [RFC3550] is used with RTP to monitor quality of service and convey information about RTP session participants. A sender of RTCP packets that also sends RTP packets (i.e., originates an RTP stream) should use the same DSCP marking for both types of packets. If an RTCP sender doesn't send any RTP packets, it should mark its RTCP packets with the DSCP that it would use if it did send RTP packets with media similar to the RTP traffic that it receives. If the RTCP sender uses or would use multiple DSCPs that differ only in drop precedence for RTP, then it should use the DSCP with the least likelihood of drop for RTCP to increase the likelihood of RTCP packet delivery.

If the SDP bundle extension [SDP-BUNDLE] is used to negotiate sending multiple types of media in a single RTP session, then receivers will send separate RTCP reports for each type of media, using a separate SSRC for each media type; each RTCP report should be marked with the DSCP corresponding to the type of media handled by the reporting SSRC.

This guidance may result in different DSCP markings for RTP streams and RTCP receiver reports about those RTP streams. The resulting variation in network QoS treatment by traffic direction is necessary to obtain representative round-trip time (RTT) estimates that correspond to the media path RTT, which may differ from the transport protocol RTT. RTCP receiver reports may be relatively infrequent, and hence the resulting RTT estimates are of limited utility for transport protocol congestion control (although those RTT estimates have other important uses; see [RFC3550]). For this reason, it is important that RTCP receiver reports sent by an SSRC receive the same network QoS treatment as the RTP stream being sent by that SSRC.

6. Guidelines

The only use of multiple standardized PHBs and DSCPs that does not enable network reordering among packets marked with different DSCPs is use of PHBs within a single AF class. All other uses of multiple PHBs and/or the class selector DSCPs enable network reordering of packets that are marked with different DSCPs. Based on this and the foregoing discussion, the guidelines in this section apply to use of Diffserv with real-time communications.

Applications and other traffic sources (including RTP SSRCs):

  • Should limit use of DSCPs within a single RTP stream to those whose corresponding PHBs do not enable packet reordering. If this is not done, significant network reordering may overwhelm implementation assumptions about reordering limits, e.g., jitter buffer size, causing poor user experiences (see Section 5.2). This guideline applies to all of the RTP streams that are within the scope of a common (coupled) congestion controller when that controller does not use per-RTP-stream measurements for bottleneck detection.
  • Should use a single DSCP for RTCP packets, which should be a DSCP used for RTP packets that are or would be sent by that SSRC (see Section 5.4).
  • Should use a single DSCP for all packets within a reliable transport protocol session (e.g., TCP connection, SCTP association) or DCCP connection (see Sections 5.1 and 5.3). For SCTP, this requirement applies across the entire SCTP association, and not just to individual streams within an association. When TURN selects TCP for NAT/FW traversal, this guideline applies to all traffic multiplexed onto that TCP connection, in contrast to use of UDP for NAT/FW traversal.
  • May use different DSCPs whose corresponding PHBs enable reordering within a single UDP or UDP-Lite 5-tuple, subject to the above constraints. The service differentiation provided by such usage is unreliable, as it may be removed or changed by DSCP remarking at network boundaries as described in Section 3.2 above.
  • Cannot rely on end-to-end preservation of DSCPs as network node remarking can change DSCPs and remove drop precedence distinctions (see Section 3.2). For example, if a source uses drop precedence distinctions within an AF class to identify different types of video frames, using those DSCP values at the receiver to identify frame type is inherently unreliable.
  • Should limit use of the CS1 codepoint to traffic for which best effort forwarding is acceptable, as network support for use of CS1 to select a "less than best-effort" PHB is inconsistent. Further, some networks may treat CS1 as providing "better than best-effort" forwarding behavior.

There is no guidance in this memo on how network operators should differentiate traffic. Networks may support all of the PHBs discussed herein, classify EF and AFxx traffic identically, or even remark all traffic to best effort at some ingress points. Nonetheless, it is useful for applications and other traffic sources to provide finer granularity DSCP marking on packets for the benefit of networks that offer QoS service differentiation. A specific example is that traffic originating from a browser may benefit from QoS service differentiation in within-building and residential access networks, even if the DSCP marking is subsequently removed or simplified. This is because such networks and the boundaries between them are likely traffic bottleneck locations (e.g., due to customer aggregation onto common links and/or speed differences among links used by the same traffic).

7. Security Considerations

The security considerations for all of the technologies discussed in this memo apply; in particular, see the security considerations for RTP in [RFC3550] and Diffserv in [RFC2474] and [RFC2475].

Multiplexing of multiple protocols onto a single UDP 5-tuple via encapsulation has implications for network functionality that monitors or inspects individual protocol flows, e.g., firewalls and traffic monitoring systems. When implementations of such functionality lack visibility into encapsulated traffic (likely for many current implementations), it may be difficult or impossible to apply network security policy and associated controls at a finer granularity than the overall UDP 5-tuple.

Use of multiple DSCPs that enable reordering within an overall real- time communication interaction enlarges the set of network forwarding resources used by that interaction, thereby increasing exposure to resource depletion or failure, independent of whether the underlying cause is benign or malicious. This represents an increase in the effective attack surface of the interaction and is a consideration in selecting an appropriate degree of QoS differentiation among the components of the real-time communication interaction. See Section of [RFC6274] for related discussion of DSCP security considerations.

Use of multiple DSCPs to provide differentiated QoS service may reveal information about the encrypted traffic to which different service levels are provided. For example, DSCP-based identification of RTP streams combined with packet frequency and packet size could reveal the type or nature of the encrypted source streams. The IP header used for forwarding has to be unencrypted for obvious reasons, and the DSCP likewise has to be unencrypted to enable different IP forwarding behaviors to be applied to different packets. The nature of encrypted traffic components can be disguised via encrypted dummy data padding and encrypted dummy packets, e.g., see the discussion of traffic flow confidentiality in [RFC4303]. Encrypted dummy packets could even be added in a fashion that an observer of the overall encrypted traffic might mistake for another encrypted RTP stream.

8. References

8.1. Normative References

   [RFC768]   Postel, J., "User Datagram Protocol", STD 6, RFC 768,
              DOI 10.17487/RFC0768, August 1980,
   [RFC793]   Postel, J., "Transmission Control Protocol", STD 7,
              RFC 793, DOI 10.17487/RFC0793, September 1981,
   [RFC2474]  Nichols, K., Blake, S., Baker, F., and D. Black,
              "Definition of the Differentiated Services Field (DS
              Field) in the IPv4 and IPv6 Headers", RFC 2474,
              DOI 10.17487/RFC2474, December 1998,
   [RFC2475]  Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z.,
              and W. Weiss, "An Architecture for Differentiated
              Services", RFC 2475, DOI 10.17487/RFC2475, December 1998,
   [RFC2597]  Heinanen, J., Baker, F., Weiss, W., and J. Wroclawski,
              "Assured Forwarding PHB Group", RFC 2597,
              DOI 10.17487/RFC2597, June 1999,
   [RFC3246]  Davie, B., Charny, A., Bennet, J., Benson, K., Le Boudec,
              J., Courtney, W., Davari, S., Firoiu, V., and D.
              Stiliadis, "An Expedited Forwarding PHB (Per-Hop
              Behavior)", RFC 3246, DOI 10.17487/RFC3246, March 2002,
   [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, <>.
   [RFC3662]  Bless, R., Nichols, K., and K. Wehrle, "A Lower Effort
              Per-Domain Behavior (PDB) for Differentiated Services",
              RFC 3662, DOI 10.17487/RFC3662, December 2003,
   [RFC3828]  Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., Ed.,
              and G. Fairhurst, Ed., "The Lightweight User Datagram
              Protocol (UDP-Lite)", RFC 3828, DOI 10.17487/RFC3828, July
              2004, <>.
   [RFC4340]  Kohler, E., Handley, M., and S. Floyd, "Datagram
              Congestion Control Protocol (DCCP)", RFC 4340,
              DOI 10.17487/RFC4340, March 2006,
   [RFC4960]  Stewart, R., Ed., "Stream Control Transmission Protocol",
              RFC 4960, DOI 10.17487/RFC4960, September 2007,
   [RFC5405]  Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
              for Application Designers", BCP 145, RFC 5405,
              DOI 10.17487/RFC5405, November 2008,
   [RFC5865]  Baker, F., Polk, J., and M. Dolly, "A Differentiated
              Services Code Point (DSCP) for Capacity-Admitted Traffic",
              RFC 5865, DOI 10.17487/RFC5865, May 2010,
   [RFC6951]  Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream
              Control Transmission Protocol (SCTP) Packets for End-Host
              to End-Host Communication", RFC 6951,
              DOI 10.17487/RFC6951, May 2013,
   [RFC7656]  Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
              B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
              for the Real-Time Transport Protocol (RTP) Sources",
              RFC 7656, DOI 10.17487/RFC7656, November 2015,

8.2. Informative References


Welzl, M., Islam, S., and S. Gjessing, "Coupled congestion

control for RTP media", Work in Progress,

draft-welzl-rmcat-coupled-cc-05, June 2015.


              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
              Channels", Work in Progress, draft-ietf-rtcweb-data-
              channel-13, January 2015.


              Geib, R., Ed. and D. Black, "Diffserv interconnection
              classes and practice", Work in Progress, draft-ietf-tsvwg-
              diffserv-intercon-03, October 2015.
   [H.221]    ITU-T, "Frame structure for a 64 to 1920 kbit/s channel in
              audiovisual teleservices", Recommendation H.221, March
   [H.264]    ITU-T, "Advanced video coding for generic audiovisual
              services", Recommendation H.264, February 2014.
   [RFC2697]  Heinanen, J. and R. Guerin, "A Single Rate Three Color
              Marker", RFC 2697, DOI 10.17487/RFC2697, September 1999,
   [RFC2698]  Heinanen, J. and R. Guerin, "A Two Rate Three Color
              Marker", RFC 2698, DOI 10.17487/RFC2698, September 1999,
   [RFC2914]  Floyd, S., "Congestion Control Principles", BCP 41,
              RFC 2914, DOI 10.17487/RFC2914, September 2000,
   [RFC3168]  Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
              of Explicit Congestion Notification (ECN) to IP",
              RFC 3168, DOI 10.17487/RFC3168, September 2001,
   [RFC3270]  Le Faucheur, F., Wu, L., Davie, B., Davari, S., Vaananen,
              P., Krishnan, R., Cheval, P., and J. Heinanen, "Multi-
              Protocol Label Switching (MPLS) Support of Differentiated
              Services", RFC 3270, DOI 10.17487/RFC3270, May 2002,
   [RFC4103]  Hellstrom, G. and P. Jones, "RTP Payload for Text
              Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,
   [RFC4303]  Kent, S., "IP Encapsulating Security Payload (ESP)",
              RFC 4303, DOI 10.17487/RFC4303, December 2005,
   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
              July 2006, <>.
   [RFC4594]  Babiarz, J., Chan, K., and F. Baker, "Configuration
              Guidelines for DiffServ Service Classes", RFC 4594,
              DOI 10.17487/RFC4594, August 2006,
   [RFC5109]  Li, A., Ed., "RTP Payload Format for Generic Forward Error
              Correction", RFC 5109, DOI 10.17487/RFC5109, December
              2007, <>.
   [RFC5127]  Chan, K., Babiarz, J., and F. Baker, "Aggregation of
              Diffserv Service Classes", RFC 5127, DOI 10.17487/RFC5127,
              February 2008, <>.
   [RFC5129]  Davie, B., Briscoe, B., and J. Tay, "Explicit Congestion
              Marking in MPLS", RFC 5129, DOI 10.17487/RFC5129, January
              2008, <>.
   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              DOI 10.17487/RFC5245, April 2010,
   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              DOI 10.17487/RFC5389, October 2008,
   [RFC5462]  Andersson, L. and R. Asati, "Multiprotocol Label Switching
              (MPLS) Label Stack Entry: "EXP" Field Renamed to "Traffic
              Class" Field", RFC 5462, DOI 10.17487/RFC5462, February
              2009, <>.
   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761,
              DOI 10.17487/RFC5761, April 2010,
   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for the Secure
              Real-time Transport Protocol (SRTP)", RFC 5764,
              DOI 10.17487/RFC5764, May 2010,
   [RFC5766]  Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
              Relays around NAT (TURN): Relay Extensions to Session
              Traversal Utilities for NAT (STUN)", RFC 5766,
              DOI 10.17487/RFC5766, April 2010,
   [RFC6062]  Perreault, S., Ed. and J. Rosenberg, "Traversal Using
              Relays around NAT (TURN) Extensions for TCP Allocations",
              RFC 6062, DOI 10.17487/RFC6062, November 2010,
   [RFC6274]  Gont, F., "Security Assessment of the Internet Protocol
              Version 4", RFC 6274, DOI 10.17487/RFC6274, July 2011,
   [RFC6386]  Bankoski, J., Koleszar, J., Quillio, L., Salonen, J.,
              Wilkins, P., and Y. Xu, "VP8 Data Format and Decoding
              Guide", RFC 6386, DOI 10.17487/RFC6386, November 2011,
   [RFC6437]  Amante, S., Carpenter, B., Jiang, S., and J. Rajahalme,
              "IPv6 Flow Label Specification", RFC 6437,
              DOI 10.17487/RFC6437, November 2011,
   [RFC6458]  Stewart, R., Tuexen, M., Poon, K., Lei, P., and V.
              Yasevich, "Sockets API Extensions for the Stream Control
              Transmission Protocol (SCTP)", RFC 6458,
              DOI 10.17487/RFC6458, December 2011,
   [RMCAT-CC] Jesup, R. and Z. Sarker, "Congestion Control Requirements
              for Interactive Real-Time Media", Work in Progress,
              draft-ietf-rmcat-cc-requirements-09, December 2014.


              Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
              Communication (WebRTC): Media Transport and Use of RTP",
              Work in Progress, draft-ietf-rtcweb-rtp-usage-25, June


              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", Work in Progress, draft-ietf-
              mmusic-sdp-bundle-negotiation-23, July 2015.


              Petit-Huguenin, M. and G. Salgueiro, "Multiplexing Scheme
              Updates for Secure Real-time Transport Protocol (SRTP)
              Extension for Datagram Transport Layer Security (DTLS)",
              Work in Progress, draft-petithuguenin-avtcore-rfc5764-mux-
              fixes-02, March 2015.


              Burnett, D., Bergkvist, A., Jennings, C., and A.
              Narayanan, "Media Capture and Streams", World Wide Web
              Consortium Recommendation WD-mediacapture-streams-
              20130903, September 2013, <


Alvestrand, H., "Overview: Real Time Protocols for

Browser-based Applications", Work in Progress,

draft-ietf-rtcweb-overview-14, June 2015.


              Alvestrand, H., "Transports for WebRTC", Work in
              Progress, draft-ietf-rtcweb-transports-10, October 2015.


This memo is the result of many conversations that have occurred within the DART working group and other working groups in the RAI and Transport areas. Many thanks to Aamer Akhter, Harald Alvestrand, Fred Baker, Richard Barnes, Erin Bournival, Ben Campbell, Brian Carpenter, Spencer Dawkins, Keith Drage, Gorry Fairhurst, Ruediger Geib, Cullen Jennings, Jonathan Lennox, Karen Nielsen, Colin Perkins, James Polk, Robert Sparks, Tina Tsou, Michael Welzl, Dan York, and the DART WG participants for their reviews and comments.

Authors' Addresses

   David Black (editor)
   176 South Street
   Hopkinton, MA  01748
   United States
   Phone: +1 508 293-7953
   Paul Jones
   7025 Kit Creek Road
   Research Triangle Park, NC  27502
   United States
   Phone: +1 919 476 2048