J. Chesterfield University of Cambridge E. Schooler Internet Draft AT&T Labs - Research Document: draft-ietf-avt-rtcpssm-04 J. Ott Tellique GmbH Expires: December 2003 29 June 2003 RTCP Extensions for Single-Source Multicast Sessions with Unicast Feedback Status of this Memo This document is an Internet-Draft and is in full conformance with all provisions of Section 10 of RFC2026. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet- Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet- Drafts as reference material or to cite them other than as work in progress. The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html. Abstract This document specifies a modification to the Real-time Transport Control Protocol (RTCP) to use unicast feedback. The proposed extension is useful for single-source multicast sessions such as Source-Specific Multicast (SSM) communication where the traditional model of many-to-many group communication is either not possible or not preferred. In addition, it can be applied to any group that might benefit from a sender-controlled summarised reporting mechanism. Table of Contents 1. Conventions and Acronyms........................................2 2. Introduction....................................................2 3. Basic Operation.................................................4 4. Definitions.....................................................5 5. Packet types....................................................6 6. Simple feedback model...........................................6 7. Sender feedback summary model...................................7 Chesterfield Internet Draft - Expires December 2002 [Page 1] RTCP with Unicast Feedback 8. Mixer/Translator issues........................................16 9. Transmission interval calculation..............................18 10. SDP Extensions................................................18 11. Security Considerations.......................................19 12. Backwards Compatibility.......................................26 13. IANA Considerations...........................................27 14. Outstanding Issues............................................29 15. References....................................................29 16. Appendix......................................................31 A Distribution Report processing at the receiver.................31 B Distribution Report creation at the source......................33 C AUTHORS ADDRESSES...............................................36 D FULL COPYRIGHT STATEMENT........................................36 1. Conventions and Acronyms The keywords MUST, MUST NOT, REQUIRED, SHALL, SHALL NOT, SHOULD, SHOULD NOT, RECOMMENDED, MAY, and OPTIONAL, when they appear in this document, are to be interpreted as described in RFC 2119. 2. Introduction The Real-time Transport Protocol (RTP) [1] provides a real-time transport mechanism suitable for unicast or multicast communication between multimedia applications. Typical uses of RTP are for real- time or near real-time group communication of audio and video data streams. An important component of the RTP protocol is the control channel, defined as the Real-Time Control Protocol (RTCP). RTCP involves the periodic transmission of control packets between group members in a session, enabling group size estimation and the distribution or calculation of session-specific information such as packet loss and round trip time to other hosts. An additional advantage of providing a control channel for a session is that a third-party session monitor can listen to the traffic to establish network conditions and to diagnose faults based on receiver locations. RTP was designed to operate in a unicast mode or in the traditional multicast mode of Any Source Multicast (ASM) group communication, where both one-to-many and many-to-many communication are supported via a common group address in the range 224.0.0.0 through 239.255.255.255. Typical intra-domain routing protocols, that is protocols that operate only within a single administrative domain, that enable such communication are the Distance Vector Multicast Routing Protocol (DVMRP) [2] or Protocol Independent Multicast (PIM) [3][4]. Typically these are used in combination with an Inter-domain routing protocol, that is to say a protocol that operates across administrative domain borders, such as the Multi-protocol extension to the Border Gateway Protocol (MBGP) [5] in combination with the Chesterfield, et al. Internet Draft - Expires December 2002 [Page 2] RTCP with Unicast Feedback Multicast Source Discovery Protocol (MSDP) [6]. Such routing protocols enable a host to join a single multicast group address and to send to or to receive data from all members in the group with no prior knowledge of the membership across any multicast-enabled portion of the internet. In order to enable such a service in the network, however, there is a great deal of complexity involved at the routing level. The many-to-many mode of communication however is not always desired by or, in some cases, even available to an application. The recent popularity of Source-Specific Multicast (SSM) is one such example where the multicast distribution channel is only available to source-to-receiver traffic. In other cases, such as large ASM groups with a single active data source and many passive receivers, it is not optimal to create at the routing level a full mesh of multicast sources just for the distribution of control packets. Thus an alternative solution is preferable. The effect of using a unidirectional broadcast topology for RTP is that it removes the ability for receivers in the group to communicate RTCP control information with all other members in the group, whether for reasons of resource economy or availability. In this draft, therefore, we define a solution to this problem. We introduce unicast feedback as a new method to distribute control information amongst all members in a multicast session. It is designed to operate under any group communication scenario (ASM or SSM). The RTP data stream protocol itself is unaffected. The basic architectural models to which the unicast feedback method could provide benefit include but are not limited to: a) SSM groups with a single sender. The proposed extensions allow SSM groups that do not have many- to-many communication capability to still receive RTP data streams and to continue to participate in the RTP control protocol, RTCP, by using multicast in the source-to-receiver Direction and unicasting receiver feedback to the source on the standard RTCP port. b) One-to-many broadcast networks. An example of such a network is a satellite network with a terrestrial low-bandwidth return channel or a broadband cable link. Unlike the SSM network, this communication architecture may have the ability for a receiver to multicast return data to the group, however, a unicast feedback mechanism is likely to be preferable for routing simplicity. c) ASM with a single sender. An SDP [16] session announcement may identify a session as having a single sender receiving unicast RTCP feedback. Receivers join the multicast group address and receive RTP and RTCP data from the source on the specified address/port combinations. However, the RTCP feedback is unicast back to the source on the RTCP port. The unicast feedback approach may help to prevent overtaxing Chesterfield, et al. Internet Draft - Expires December 2002 [Page 3] RTCP with Unicast Feedback multicast routing infrastructure that does not scale as efficiently. However, it is not more efficient than a standard multicast group RTP communication scenario, and therefore is not expected to replace the traditional mechanism. The modifications proposed in this document are intended to supplement the existing RTCP feedback mechanisms described in [1], Section 6. 3. Basic Operation This draft proposes two new methods to enable receiver feedback to all members in a session. Each involves the unicasting of RTCP packets to a source whose job it is to re-distribute the information to the members of the group. The source MUST be able to communicate with all group members in order for either mechanism to work. The content of receiver RTCP traffic will continue to include Receiver Reports (RRs) and Session Description (SDES) information since they are never active sources in the session. Additionally, Goodbye (BYE) packets and Application-defined (APP) packets may also be transmitted. The various reports may be combined into a single RTCP packet, which should not exceed the path MTU. Packets continue to be sent at a rate that is inversely proportional to the group size in order to scale the amount of traffic generated. The first method, the 'Simple Feedback Model', is a basic reflection mechanism whereby all Receiver RTCP packets are unicast to the source and subsequently forwarded by the source to all receivers on the multicast RTCP channel. The advantage of using this method is that an existing receiver implementation requires little modification in order to use it. Instead of sending reports to a multicast address, a receiver uses a unicast address to send reports to the source, yet still receives forwarded RTCP traffic on the multicast control data channel. This method also has the advantage of being backwards compatible with RTP/RTCP implementations that do not support unicast feedback to the source and that operate using the standard multicast group communication model, ASM. In a session that uses ASM, such a receiver would multicast reports to the group address and designated RTCP port as stated in [1]. The reports would be directly received by all members. In a session using SSM, the network SHOULD prevent any multicast data from the receiver being distributed further than the first hop router. Additionally, any data heard from a non-unicast capable receiver by other hosts on the same subnet SHOULD be filtered out by the host IP stack and therefore will not cause problems with respect to the calculation of the Receiver RTCP bandwidth share. The second method, the 'Sender Feedback Summary Model', is a summarised reporting scheme that provides savings in bandwidth by consolidating Receiver Reports at the source into one summary packet that is then distributed to all the receivers. The advantage of this scheme is apparent for large group sessions where the basic Chesterfield, et al. Internet Draft - Expires December 2002 [Page 4] RTCP with Unicast Feedback forwarding mechanism outlined above would create a large amount of packet replication to forward all the information to all the receivers. The basic operation of the scheme is the same as the first method; receivers send feedback via unicast to the source, and the source distributes summaries of the feedback over the multicast channel. However, the technique requires that all session members understand the new summarised packet format outlined in Section 7.1. Additionally, the summarised scheme provides an optional mechanism to send distribution information or histograms about the feedback data reported by the whole group. Potential uses for the compilation of distribution information are addressed in Section 7.4. To differentiate between the two reporting methods, a new SDP identifier is created and discussed in Section 10. The reporting method MUST be decided prior to the start of the session. A distribution source MUST NOT change the method during a session. 4. Definitions Distribution Source: In an SSM context, only one source distributes RTP data and redistributes RTCP information to all receivers. That source is called the distribution source. In order for unicast feedback to work, there MUST be only one session distribution source for any subset of receivers to which RTCP feedback is directed. Note that heterogeneous networks comprised of ASM multiple-sender groups, unicast-only clients and/or SSM single-sender receiver groups MAY be connected via translators or mixers to create a single-source group (see Section 9 for details). RTP and RTCP Channels: The data distributed from the source to the receivers is referred to as the RTP channel and the control information the RTCP channel. With standard RTP/RTCP, these channels typically share the same multicast address but are differentiated via port numbers as specified in [1]. Unicast RTCP Feedback Target: For a session defined as having a distribution source A, on ports n and k, the unicast RTCP feedback target is the IP address of Source A on port k unless otherwise stated in the session description. See Section 10 for details on how the address is specified. SSRC: Synchronization source. A 32-bit value that uniquely identifies each member in a session. See [1] for further information. Report blocks: RTCP [1] encourages the stacking of multiple report blocks in Sender and Receiver Report packets. As a result, a variable size feedback packet is created and sent from one source that pertains to multiple other sources in the group. Report block is the standard terminology for an RTCP reception report and multiple report blocks may be sent in the same packet. Chesterfield, et al. Internet Draft - Expires December 2002 [Page 5] RTCP with Unicast Feedback 5. Packet types The RTCP packet types defined in [1] are: Type Description Payload number SR Sender Report 200 RR Receiver Report 201 SDES Source Description 202 BYE Goodbye 203 APP Application-Defined 204 RTPFB Generic RTP feedback 205 PSFB Payload-specific feedback 206 XR RTCP Extension 207 This document defines one further RTCP packet format: type description Payload number ---------------------------------------------------------------- RSI Receiver Summary Information 208 Within the Receiver Summary Information packet, various types of distribution data may be reported, each of which requires a distribution type identifier for report blocks. In addition, other information may be reported, also encapsulated in separate report blocks. The sub-types identifying the report blocks are: Sub-block Type Description Identifier number ------------------------------------------------------------------ IPv4 Address IPv4 Unicast Feedback address 0 IPv6 Address IPv6 Unicast Feedback address 1 DNS name DNS name for Unicast Feedback 2 - - reserved - 3 Jitter Jitter distribution 4 RTT Round trip time distribution 5 Cumulative loss Cumulative loss distribution 6 Loss Loss distribution 7 Collisions SSRC collision list 8 BYE BYE list 9 Stats General statistics 10 Receiver BW RTCP Receiver Bandwidth 11 - - reserved - 12 - 255 6. Simple feedback model 6.1 Packet Formats The simple feedback model uses the same packet types as standard RTCP feedback described in [1]. Receivers still generate Receiver Reports with information on the quality of the stream received from the source. The distribution source still must create Sender Reports Chesterfield, et al. Internet Draft - Expires December 2002 [Page 6] RTCP with Unicast Feedback that include timestamp information for stream synchronisation and round trip time calculation. Both senders and receivers are required to send SDES packets as outlined in [1]. The rules for generating BYE and APP packets as outlined in [1] also apply. 6.2 Distribution Source behaviour For the simple feedback model, the source provides a basic packet reflection mechanism. It is the default behaviour for any distribution source and is the minimum requirement for acting as a source to a group of receivers using unicast RTCP feedback. In this model, the source MUST not stack report blocks received from different SSRCs into one packet for retransmission to the group. Every RTCP packet from each receiver MUST be reflected individually. The source MUST listen for unicast RTCP data sent to the RTCP port. All unicast data received on this port MUST be forwarded by the source to the group on the multicast RTCP channel. If the application can determine the destination address of an RTCP packet as being multicast, the packet MUST NOT be forwarded but processed as defined in [1]. The reflected traffic SHOULD NOT be included in the transmission interval calculation by the source. In other words, the source SHOULD NOT consider reflected packets as part of its own control data bandwidth allowance. The algorithm for computing the allowance is explained in Section 9. If an application wishes to use APP packets, it is recommended that the simple feedback model be used since it is likely that all receivers in the session will need to hear the APP specific packets. The same applies for all other RTCP packets that are not defined in the base RTP specification [1]. The decision to use the simple feedback model MUST be made in advance of the session and MUST be indicated in the SDP announcement [16]. 6.3 Receiver behaviour Receivers listen on the RTP channel for data and the RTCP channel for control. Each receiver calculates its share of the control bandwidth based on the standard rule that 75% of the RTCP bandwidth is divided equally between all unique SSRCs in the session. See Section 9 for further information on the calculation of the bandwidth allowance. When a receiver is eligible to transmit, it sends a unicast Receiver Report packet to the RTCP port of the distribution source. 7. Sender feedback summary model In the sender feedback summary model, the source is required to summarise the information received from all the Receiver Reports generated by the receivers and place the information into summary Chesterfield, et al. Internet Draft - Expires December 2002 [Page 7] RTCP with Unicast Feedback reports. The sender feedback summary model introduces a new report block format and a number of optional report block formats. The Receiver Summary Information report (RSI) is required and MUST be sent by a source if the summarised feedback mechanism is selected. Transmission of sub-report types is OPTIONAL. They MAY be appended to the RSI report block to provide more detailed information on the overall session characteristics reported by all receivers and also to convey important information such as the feedback address and reporting bandwidth. The sender MUST send at least one Receiver Summary Information packet for each reporting interval. The sender can additionally stack report blocks after the RSI packet. Each report block corresponds to the initial RSI packet and acts as an enhancement to the basic summary information required by the receivers to calculate their reporting time interval. For this reason, report blocks are not required but recommended. RSI and corresponding report blocks are sent in addition to the standard sender-issued packets, such as Sender Reports and SDES packets outlined in [1]. 7.1 Packet Formats 7.1.1 RSI: Receiver Summary Information Packet The RSI report block has a fixed header size of 4 octets followed by a variable length report: 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |V=2|P|reserved | PT=RSI=208 | length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC/CSRC | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Timestamp | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Group size | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ : optional report blocks : +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ The RSI packet includes the following fields: SSRC: 32 bits The SSRC of the distribution source. Timestamp: 32 bits The time the packet was sent. This is an unsigned integer value displayed in NTP timestamp units to enable detection of duplicate packets, reordering and to provide a chronological profile of the feedback reports. Chesterfield, et al. Internet Draft - Expires December 2002 [Page 8] RTCP with Unicast Feedback Group size: 32 bits This field provides the sender's view of the number of receivers in a session. This MUST include the sender itself and any other senders potentially connected to the session, e.g., via a mixer/translator gateway. The group size is calculated according to the rules outlined in [1]. 7.1.2 Optional Report Block Types For RSI reports, this document also introduces a report block format specific to the RSI packet. The report blocks are appended to the RSI packet using the following generic format. All report blocks MUST be 32-bit aligned. 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | RBT | Length | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ RBT-specific data + | | : : +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ RBT: 8 bits Sub-Block Type. The sub-block type identifier. The values for the sub-block types are defined in section 5. Length: 8 bits The length of the sub-report in 32-bit words. RBT-specific data: octets This field may contain type-specific information based upon the SBT value. 7.1.3 Generic Sub-Block Fields For the sub-blocks that convey distributions of values (Loss, Jitter, RTT, Cumulative Loss), a flexible 'data bucket' style report is used. This divides the data set into variable size buckets that are read based upon the guide fields at the head of the report block. 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | RBT | Length | NDB | MF | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Minimum Distribution Value | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Maximum Distribution Value | Chesterfield, et al. Internet Draft - Expires December 2002 [Page 9] RTCP with Unicast Feedback +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | Distribution Buckets | | ... | | ... | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ The RBT and Length fields are calculated as explained in section 7.1.2. Number of distribution buckets (NDB): 12 bits The number of distribution buckets of the data. The size of the bucket can be calculated using the formula, number of bits equals ((length * 4) - 12)*8/NDB. The calculation is based upon the length of the whole packet in octets (length field * 4) minus the header of 12 octets. Providing 12 bits for the NDB field enables bucket sizes as small as 2 bits for a full length packet. The bucket size in bits must always be divisible by 2 to ensure proper byte alignment. A bucket size of 2 bits is fairly restrictive, however, and it is expected that larger bucket sizes will be more practical for most distributions. Multiplicative Factor (MF): 4 bits MF+1 indicates the multiplicative factor to be applied to each distribution Bucket value. Possible values are 0 - 15, indicating factors 1 - 16. Length: 8 bits The length field tells the receiver the full length of the sub- block in 32-bit words, and enables the receiver to identify the bucket size based on the calculation ((length * 4) - 12)*8/NDB. Therefore the maximum data portion of a distribution packet may be 1008 bytes, which would provide up to 4032 data buckets of length 2 bits, or 2016 data buckets of length 4 bits etc... Minimum distribution value: 32 bits The minimum distribution value, in combination with the maximum distribution value, indicates the range covered by the data bucket values. Maximum distribution value: 32 bits The maximum distribution value, in combination with the minimum distribution value, indicates the range covered by the data bucket values. The significance and range of the distribution values is defined in the individual profiles for each distribution type (DT). Distribution buckets: each bucket is((length * 4) - 12)*8/NDB bits The size and number of buckets depends upon the value of NDB and the length of the packet. In order to calculate the size of the bucket, the formula ((length * 4) - 12)*8/NDB should be used. This indicates the division of the data space and the size of each data point in bits. Each value must be multiplied by the multiplicative factor. Chesterfield, et al. Internet Draft - Expires December 2002 [Page 10] RTCP with Unicast Feedback Interpretation of the minimum, maximum and distribution values in the report block are profile-specific and are addressed individually in the sections below. The size of the report block is variable, as indicated by the packet length field. 7.1.4 Loss report block A loss report block indicates the distribution of losses as reported by the receivers to the distribution source. Values are expressed as a fixed-point number with the binary point at the left edge of the field. The sub-block type (SBT) is 3. Valid results for the minimum distribution value field are 0 - 254. Similarly, valid results for the maximum distribution value field are 1 - 255. The minimum distribution value MUST always be less than the maximum. For examples on processing summarised loss report blocks, see the Appendix. 7.1.5 Jitter report block A jitter report block indicates the distribution of the estimated statistical variance of the RTP data packet interarrival time reported by the receivers to the distribution source. See [1] for details on how the values are calculated and the relevance of the jitter results. Jitter values are measured in timestamp units and expressed as unsigned integers. The minimum distribution value must always be less than the maximum. The sub-block type (SBT) is 4. 7.1.6 Round Trip Time report block A round trip time report indicates the distribution of round trip times from the distribution source to the receivers. The distribution source is the only member of the group capable of calculating the round trip time to any other members since it is the only sender in the group. The sender has the option of distributing these round trip time estimations to the whole group, uses of which are described in Section 7.4. Round trip times are measured in timestamp units and expressed as unsigned integers. The multiplicative factor can be used to reduce the number of bits required to represent the values. The minimum distribution value MUST always be less than the maximum. The sub-block type (SBT) is 5. 7.1.7 Cumulative Loss report block The cumulative loss field, in contrast to the Average Fraction Lost field, in a Receiver Report [1] is intended to provide some historical perspective on the session performance. The distribution is provided as a percentage figure based on the long term accumulation of values. The sender must maintain a record of the Cumulative number lost and Extended Highest Sequence number received Chesterfield, et al. Internet Draft - Expires December 2002 [Page 11] RTCP with Unicast Feedback as reported by each receiver, ideally the recorded values are from the first report received. Future calculations are then based on (the new cumulative loss value - the recorded value) / (new extended highest sequence number - recorded sequence number) * 100. The sub- block type (SBT) is 6. 7.1.8 Feedback Address Target report block 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | RBT={0,1,2} | Length | Port | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | | : Address : | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Length: 8 bits The length of the report block in 32-bit words. For an IPv4 address this should be 2 (e.g. Total length = 4 + 4 = 2*4 Octets). For an IPv6 address this should be 5). For a DNS name, the length field indicates the padded number of 32-bit words making up the string plus 1. Port: 2 octets (optional) The port number to direct reports to. If port==0, the port number MUST be ignored. Address: 4 octets (IPv4), 16 octets (IPv6), or n octets (DNS name) The address to which receivers send feedback reports. For IPv4 and IPv6 fixed-length address fields are used. A DNS name is an arbitrary length string that is padded with null bytes to the next 32 bit boundary. The string is UTF-8 encoded (RFC 2279). For IPv4, SBT=0. For IPv6, SBT=1. And for the DNS name for unicast feedback, SBT=2. 7.1.9 Collisions report block 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | RBT=8 | Length | Reserved | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | | : Collision SSRC : | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Collision SSRC: n x 32 bits Chesterfield, et al. Internet Draft - Expires December 2002 [Page 12] RTCP with Unicast Feedback The final fields in the packet are used to identify any SSRCs that are duplicated within the group. Usually this is handled by the hosts upon detection of the same SSRC, however since receivers in an SSM context no longer have a global view of the session, the collision algorithm is handled by the source. SSRCs that collide are listed in the packet. Each Collision SSRC is included repeatedly in Collision report blocks and sent at least five times. If more Collision SSRCs need to be reported than fit into an MTU, the reporting is done in a round robin fashion so that all Collision SSRCs have been reported once before the second round of reporting starts. On receipt of the packet, receiver(s) MUST detect the collision and select another SSRC, If the collision pertains to them. 7.1.10 BYE report block 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | RBT=9 | Length | Reserved | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | | : Leaving SSRC : | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Leaving SSRC: n x 32 bits The final fields in the packet are used to identify any SSRCs from which a BYE message has been received. Each Leaving SSRC is included repeatedly in BYE list report blocks and sent at least five times. If more Leaving SSRCs need to be reported than fit into an MTU, the reporting is done in a round robin fashion so that all Leaving SSRCs have been reported once before the second round of reporting starts. Receivers of BYE report blocks remove the corresponding SSRCs from their list of session members. 7.1.11 General Statistics report block 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | RBT=10 | Length | Reserved | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | AFL | HCNL | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Average interarrival jitter | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Average fraction lost (AFL): 8 bits The average fraction lost indicated by Receiver Reports forwarded to this source, expressed as a fixed point number with the binary Chesterfield, et al. Internet Draft - Expires December 2002 [Page 13] RTCP with Unicast Feedback point at the left edge of the field. A value of all '1's indicates that the AFL field is not provided. Highest cumulative number of packets lost (HCNL): 24 bits Highest 'cumulative number of packets lost' value out of all RTCP RR packets since the last RSI from any of the receivers. A value of all '1's indicates that the HCNL field is not provided. Average interarrival jitter: 32 bits Average 'interarrival jitter' value out of all RTCP RR packets since the last RSI from the receiver set. A value of all '1's indicates that this field is not provided. 7.1.12 RTCP Bandwidth indication block 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | RBT=11 | Length |S|R| Reserved | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | RTCP Bandwidth | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Sender (S): 1 bit The contained bandwidth value applies to all RTCP senders. Receivers (R): 1 bit The contained bandwidth value applies to all RTCP receivers. Reserved: 14 bits MUST be set to zero upon transmission and ignored upon reception. RTCP Bandwidth: 32 bits Indicates the maximum bandwidth allocated to any single receiver for sending RTCP data relating to the session. This is a fraction value indicating a percentage of the session bandwidth, expressed as a fixed-point number with the binary point at the left edge of the field. Only the sender bit or the receiver bit MAY be set to 1 at the same time. 7.2 Distribution Source behaviour The source is responsible for accepting RTCP packets from the receivers, interpreting and storing the per-receiver information as defined in [1]. The importance of this is apparent when creating the RSI and sub-block reports, since incorrect information can have serious implications. Section 11 addresses the security risks in detail. Chesterfield, et al. Internet Draft - Expires December 2002 [Page 14] RTCP with Unicast Feedback Either the group size or the Receiver RTCP Bandwidth fields MUST be included. A sender has the option of masking the group size by setting it to a value of choice, however the receiver bandwidth field MUST be included. If both are included, the bandwidth field MUST override the Session bandwidth/group size calculation as outlined in [1]. The Receiver RTCP Bandwidth field therefore provides the source with the capability to control the amount of feedback from the receivers and MAY be increased or decreased based on the requirements of the source. Regardless of the value selected by the source for the Receiver RTCP Bandwidth field, the source MUST continue to forward Sender Reports and RSI packets at the rate allowed by its bandwidth allocation. See Section 9 for further details about the frequency of reports. In the event that the source receives a BYE notification from any number of receivers, it MUST forward a BYE summary report listing the SSRC or SSRCs that are leaving. The announcement MUST be made at least 5 times. If there are more SSRCs than can be listed in a BYE summary within the space available, the BYE SSRC list MUST be announced in a round-robin fashion, until all SSRCs have been announced at least 5 times. The source MUST NOT adjust the group size estimator value, either the group size or RTCP bandwidth fields, to reflect the change until the relevant SSRC has been announced at least 5 times. See Section 11 for further explanation of this. In order to identify SSRC collisions, the source is responsible for maintaining a record of each SSRC and the corresponding CNAME within at least one reporting interval by the group in order to differentiate between clients. It is RECOMMENDED that an updated list of more than one interval be maintained to increase accuracy. This mechanism should prevent the possibility of collisions since the combination of SSRC and CNAME should be unique if the CNAME is generated correctly. If collisions are not detected, the source will have an inaccurate impression of the group size. Since the statistical probability is very low that collisions will both occur and be undetectable, this should not be a significant concern. In particular, the clients would have to randomly select the same SSRC and have the same username + IP address (e.g., using private address space behind a NAT router). For the optional report blocks, the source MAY decide which are the most significant feedback values to convey and MAY choose not to include any. The packet format provides flexibility in the amount of detail conveyed by the data points. There is a trade-off between the granularity of the data and the accuracy based on the factorisation values, the number of buckets and the min and max values. In order to focus on a particular region of a distribution, the source can adjust the minimum and maximum values and either increase the number of buckets and possibly the factorisation, or decrease the number of buckets and provide more accurate values. See Appendix B for detailed examples on how to convey a summary of RTCP Receiver Reports as RSI report block information. Chesterfield, et al. Internet Draft - Expires December 2002 [Page 15] RTCP with Unicast Feedback The results should correspond as near as possible to the values received during the interval since the last report. The source may stack as many report blocks as required in order to convey different distributions. If the distribution size exceeds the largest packet length (1008 bytes data portion), more packets may be stacked with additional information up to the MTU of the session. 7.3 Receiver behaviour The receiver MUST process RSI packets and adapt session parameters such as the RTCP bandwidth based on the information received. The receiver no longer has a global view of the session, and will therefore be unable to receive information from individual receivers aside from itself. However, the information portrayed by the source can be extremely detailed, providing the receiver with an accurate view of the session quality overall, without the processing overhead associated with listening to and analysing all Receiver Reports. The SSRC collision list MUST be checked against the SSRC selected by the receiver to ensure there are no collisions. The group size value provides the receiver with the data necessary to calculate its share of the RTCP bandwidth. The Receiver RTCP Bandwidth field may override this value. The Receiver RTCP Bandwidth field provides the source with the capability to control the amount of feedback from the receivers. The BYE SSRC summarisation list MUST be checked for any entries corresponding to the receiver's own SSRC. In the event that a BYE is incorrectly advertised for a receiver that is still active, the standard process for reselecting an SSRC in the event of a collision must be initiated. See section 11 for further explanation. The receiver can use the summarised data as desired. This data is most useful in providing the receiver with a more global view of the conditions experienced by other receivers, and enables the client to place itself within the distribution and establish the extent to which its reported conditions correspond to the group reports as a whole. Appendix A provides further information and examples of data processing at the receiver. The receiver SHOULD assume that any report blocks in the same packet correspond to the same data set received by the source during the last reporting time interval. This applies to packets with multiple blocks, where each block conveys a different range of values. 8. Mixer/Translator issues The original RTP specification allows a session to use mixers and translators which help to connect heterogeneous networks into one session. There are a number of issues, however, which are raised by the unicast feedback model proposed in this document. The term 'mixer' refers to devices that provide data stream multiplexing Chesterfield, et al. Internet Draft - Expires December 2002 [Page 16] RTCP with Unicast Feedback where multiple sources are combined into one stream. Conversely, a translator does not multiplex streams, but simply acts as a bridge between two distribution mechanisms, e.g., a unicast-to-multicast network translator. Since the issues raised by this draft apply equally to either a mixer or translator, they are referred to from this point onwards as mixer-translator devices. A mixer-translator between distribution networks in a session must ensure that all members in the session receive all the relevant traffic to enable the usual operation by the clients. A typical use may be to connect an older implementation of an RTP client with an SSM distribution network, where the client is not capable of unicasting feedback to the source. In this instance the mixer- translator must join the session on behalf of the client and send and receive traffic from the session to the client. Certain hybrid scenarios may have different requirements. 8.1 Use of a mixer-translator The mixer-translator MUST adhere to the SDP description [16] for the single source session (Section 10) and use the feedback mechanism indicated. Receivers SHOULD be aware that by introducing a mixer- translator into the session, more than one source may be active in a session since the mixer-translator may be forwarding traffic from either multiple unicast sources or from an ASM session to the SSM receivers. Receivers SHOULD still forward unicast RTCP reports in the usual manner to the distribution source, which in this case would be the mixer-translator itself. It is RECOMMENDED that the simple packet reflection mechanism be used under these circumstances since attempting to coordinate RSI + summarisation reporting between more than one source may be complicated unless the mixer-translator is capable of undertaking the summarisation itself. 8.2 Encryption and Authentication issues Encryption and security issues are discussed in detail in Section 11. A mixer-translator MUST be able to follow the same security policy as the client in order to unicast forward RTCP feedback to the source, and it therefore MUST be able to apply the same authentication and/or encryption policy required for the session. Transparent bridging, where the mixer-translator is not acting as the distribution source, and subsequent unicast feedback to the source is only allowed if the mixer-translator can conduct the same source authentication as required by the receivers. A translator may forward unicast packets on behalf of a client, but SHOULD NOT translate between multicast to unicast flows towards the source without authenticating the source of the feedback address information. Chesterfield, et al. Internet Draft - Expires December 2002 [Page 17] RTCP with Unicast Feedback 9. Transmission interval calculation The Control Traffic Bandwidth referred to in [1] is an arbitrary amount which is intended to be supplied by a session management application (e.g., [9]) or decided based upon the bandwidth of a single sender in a session. A receiver MUST calculate the number of other members in a session based upon either its own SSRC count determined by the number of forwarded Receiver Reports it receives, from the group size field or the RTCP bandwidth field in a control report from a sender. The RTCP transmission Interval calculation remains the same as in the original RTP specification [1]. In the original specification, the senders are allocated 1/4 of the control traffic bandwidth if they number 25% or less than the group size. Otherwise the allocation for senders is the percentage of senders to group size. The remaining bandwidth is allocated to the receivers to be divided evenly amongst them. The source should calculate the transmission interval for RSI and summarisation packets out of its 1/4 of the control traffic bandwidth with a minimum transmission interval of 5 seconds. 10. SDP Extensions The Session Description Protocol (SDP) [16] is used as a means to describe media sessions in terms of their transport addresses, codecs, and other attributes. Providing RTCP feedback via unicast as specified in this document constitutes another session parameter needed in the session description. Similarly, other single-source multicast RTCP feedback parameters need to be provided, such as the summarisation mode at the sender and the target unicast address to which to send feedback information. This section defines the SDP parameters that are needed by the proposed mechanisms in this draft (and that also need to be registered with IANA). 10.1 SSM RTCP Session Identification A new session level attributes MUST be used to indicate the use of unicast instead of multicast feedback: "rtcp:unicast". This attribute uses one additional parameter to specify the mode of operation. rtcp:unicast reflection -- MUST be used to indicate packet reflection by the RTCP target (without further processing). rtcp:unicast rsi -- MUST be used to indicate the "Receiver Summary Information" mode of operation. 10.2 SSM Source Specification Chesterfield, et al. Internet Draft - Expires December 2002 [Page 18] RTCP with Unicast Feedback In addition, in a Source-Specific Multicast RTCP session, the sender(s) need to be indicated for both source-specific joins to the multicast group, as well as for addressing unicast RTCP packets on the backchannel from receivers to the source. This is done following the proposal for SDP source filters documented in draft-ietf-mmusic-sdp-srcfilter-05.txt [15]. From this specification, only the inclusion mode ("a=source-filter:incl") MUST be used for SSM RTCP. There SHOULD be exactly one "a=source-filter:incl" attribute listing the address of the sender. The RTCP port MUST be derived from the m= line of the media description. An optional alternative feedback address may be supplied using an attribute such as a=rtcp: IN IP4 192.168.1.1. 11. Security Considerations The level of security provided by the current RTP/RTCP model MUST NOT be diminished by the introduction of unicast feedback to the source. This section presents an analysis of the security weaknesses introduced by the feedback mechanism, identifying the potential threats and specifying the measures that MUST be adopted to address the issues. Any Suggestions on increasing the level of security provided to RTP sessions above the current standard are RECOMMENDED but OPTIONAL. 11.1 Assumptions RTP/RTCP is a protocol for carrying real-time multimedia traffic, and therefore one of the main considerations for any security solution must be to maintain as low an overhead as possible in order to limit processing constraints. This includes the consideration of overhead for different applications and types of cryptographic operations, as well as considerations for deploying or creating security infrastructure for large groups. The distribution of session parameters, typically using type information through SAP, email or the web is beyond the scope of this document. It is recommended, however, that the method used should employ adequate security measures to ensure the integrity and authenticity of the information. For the purposes of this analysis, it is assumed that the information has already been securely distributed out-of-band. This work is in progress within the Gsec working group, which is addressing the security implications of distributing session announcements. In practice, the multicast and group distribution mechanism, e.g., the SSM routing tree, is not immune to source IP address spoofing or traffic snooping. All security weaknesses are therefore addressed from a transport level perspective or above. Chesterfield, et al. Internet Draft - Expires December 2002 [Page 19] RTCP with Unicast Feedback 11.2 Security threats Attacks on this architecture may take a variety of forms, and in order to identify the security weaknesses, it is important to address these individually. a) Denial of Service A major area of concern would be a denial of service attack. Due to the nature of the communication architecture this is a situation that could be generated a number of ways by using the unicast feedback characteristic to the attackers advantage. b) Packet Forgery One potential area of attack to guard against is packet forgery. In particular, it is important to protect against the integrity of certain influential packets since compromise of certain control packets could directly affect the transmission characteristics of the whole group, however for the purposes of defining a security profile at this stage, every packet is considered equally as important. This decision may be revisited in future revisions as the requirements emerge more clearly. It is clear, however that in the case of a large group, the compromise of RTCP traffic could have serious consequences. c) Session Replay An additional concern is the potential for session recording and subsequent replay. The issue to deal with in particular in this instance is that an attacker may not actually need to understand the packet contents, but just simply have the ability to record the data stream and at a later time replay it to any receivers that are listening. This would be less effective than a direct denial of service attack, since the integrity of the original session packets would not be affected, however the feedback at that time by the receiver group to the source would be unexpected. d) Eavesdropping on a session The consequences of eavesdropping by an attacker on a session may not directly constitute a security weakness, however it might benefit other types of attack, and should therefore be considered as a potential threat. For example, an attacker might be able to use the information to perform an intelligent DoS attack. 11.3 Security properties The following security types are relevant to this draft and will be addressed in greater detail in subsequent sections. a) Data integrity Ensures that any third party has not tampered with the data received from the network, either maliciously or through a network error. This property ensures that the packet received is Chesterfield, et al. Internet Draft - Expires December 2002 [Page 20] RTCP with Unicast Feedback guaranteed to be in exactly the same condition as intended, and that what is received can be confirmed as being from an authenticated source. b) Data authenticity Ensures the origin of the message can either be uniquely identified, or in the case of group authentication can be identified as coming from a specific trusted group of sources. c) Data confidentiality Ensures that the only receiver(s) that can recover the plaintext is the intended receivers, therefore in order to restrict access only to authorized entities, confidentiality is required. It does not prevent eavesdroppers receiving the ciphertext but does guarantee that they cannot recover the plaintext. d) Replay protection This prevents an attacker from re-using the packet via simple replay without necessarily having to recover the plaintext. 11.4 Architectural Contexts In order to understand the potential weaknesses to guard against, it is necessary to divide the communication model into a number of distinct contexts. a) Source to Receiver communication The first, and perhaps most influential context to protect, is the 'downstream' communication channel from the source to the receivers. This is effectively the main controlling influence over the behaviour of the group since it determines the bandwidth allocation for each receiver and hence the rate at which the RTCP traffic is directly unicast back to the source. All traffic that is distributed over the downstream channel should be generated by a single source. Both the RTP data stream and the RTCP control data are sent over this channel. The RTCP data is indirectly influenced by the information the source has received from the whole group. This context is vulnerable to all four attacks outlined in the previous section. A denial of service attack from the source to the receivers is possible, but less of a concern since the worst case effect of sending large volumes of traffic over the distribution channel has the potential to reach every receiver, but only on a one-to-one basis, this is no different from the current multicast model where an individual source may send large volumes of traffic to a multicast group. The real danger of denial of service attacks in this context comes indirectly via compromise of the source RTCP traffic. If receivers are provided with an incorrect group size estimate or bandwidth allowance, the return traffic to the source may create a distributed DoS effect on the source. Similarly, an incorrect feedback address whether Chesterfield, et al. Internet Draft - Expires December 2002 [Page 21] RTCP with Unicast Feedback as a result of a malicious attack or by mistake, e.g., an IP address typing error, could directly create a denial of service attack on another host. An additional concern relating to Denial of Service attacks would come in the form of fake BYE packets generated by an attacker and forwarded to the source. Such an attack has potential to generate a false group size estimation by the source, potentially causing a distributed DoS effect by the receiver group. The danger of Packet forgery in the worst case may be to maliciously instigate a denial of service attack, e.g. if an attacker were capable of spoofing the source address and injecting incorrect packets into the data stream or intercepting the source RTCP traffic and modifying the fields. Other consequences of packet forgery in this context may be the compromise of data affecting the integrity of the data received both in the RTP stream itself and the RTCP data in general. The replay of a session would have the effect of recreating the receiver feedback to the source address at a time when the source is not expecting additional traffic from a potentially very large group. The consequences of this type of attack may be less effective on their own, but in combination with other attacks might be serious. Eavesdropping on the session would provide an attacker with information on the characteristics of the source to receiver traffic such as the frequency of RTCP traffic and, if unencrypted, might also provide valuable information on characteristics such as group size and transmission characteristics of the receivers back to the source in addition to enabling an attacker to listen to the media streams. In this context, the attacker might also have access to personal information carried in the SDES packets such as email, phone and full username information through traffic analysis. b) Receiver to source or gateway communication The second context to address is the return traffic from the group to the source or gateway, which for the purposes of this analysis may be considered in the same light as a distribution source. This traffic should only be RTCP type data, and should include receiver reports, SDES information and possibly Application specific packets. The effects of compromise on a single or subset of receivers is less likely to have as great an impact as the first context, however much of the responsibility for detecting compromise of the source data stream relies on the receivers. The effects of compromise of the first context with respect to critical source RTCP control information would be witnessed most seriously in the second context. A large group of receivers may unwittingly generate a distributed DoS attack on the source in Chesterfield, et al. Internet Draft - Expires December 2002 [Page 22] RTCP with Unicast Feedback the event that the integrity of the source RTCP channel has been compromised and is not detected by the individual receivers. In the event that packet forgery may occur in this context, the effect may be the introduction of false RTCP traffic and/or the creation of fake SSRC identifiers. Such an attack might slow down the overall control channel data rate, since an incorrect perception of the group size may be created. This might affect external issues such as group accounting and other as yet unknown potential uses of the distribution functionality for controlling group behaviour such as leader election based on feedback criteria. A replay attack on receiver return data to the source would have the same implications as the generation of false SSRC identifiers and RTCP traffic to the source. It is therefore equally as important to protect against compromise of any receiver contribution to the RTCP channel, as it is to ensure authenticity and freshness of the data source. Eavesdropping in this context may potentially provide an attacker with a great deal of personal information about a large group of receivers available from SDES packets. It would also provide an attacker with information on group traffic generation characteristics and parameters for calculating individual receiver bandwidth allowance. 11.5 Requirements in each context Some initial requirements to consider for each context in general are that the overhead of ensuring the security of the session should be kept as low as possible. This entails keeping the setup and communication of keys to a minimum. The nature of RTP/RTCP traffic is that sessions require real-time processing and minimal overhead for communication. This means that processing constraints imposed by the required cryptographic techniques are an important consideration in defining this security profile. Having identified the security weaknesses for each communication context, security type requirements can be addressed for each. a) The first context is concerned with denial of service attacks through possible packet forgery. The forgery may take the form of interception and modification of packets from the source, or simply injecting false packets into the distribution channel. To combat these attacks, data integrity and source authenticity of the source traffic MUST be applied. The degree of confidentiality which may be deployed is not a requirement in this context since the actual consequences of eavesdropping do not affect the operation of the protocol, however without confidentiality, access to personal and group characteristics information would be unrestricted to an external listener and it is therefore recommended. Chesterfield, et al. Internet Draft - Expires December 2002 [Page 23] RTCP with Unicast Feedback b) The second context SHOULD defend against the same kinds of attacks but is considered less important than the downstream traffic compromise. All the security weaknesses are also applicable to the current RTP/RTCP security model and therefore only recommendations are made towards protection from compromise. Data integrity is RECOMMENDED to ensure that interception and modification of an individual receiver's RTCP traffic is not accomplished. This would protect against the false influence of group control information and the potentially more serious compromise of future services provided by the distribution functionality such as leader election based on various parameters. In order to ensure security, data integrity and authenticity of receiver trafficis therefore also RECOMMENDED. The same situation applies as in the first context with respect to data confidentiality, and it is recommended that precautions should be taken to protect the privacy of the data. An additional security consideration that is not a component of this specification but has a direct influence upon the general security is the origin of the session initiation data. This involves the SDP parameters that are communicated to the members prior to the start of the session via channels such as an HTTP server, email or SAP. As it is beyond the scope of this document to place any requirements on the external communication of such information, no further analysis is included here, however it is highly RECOMMENDED that wherever possible an implementer or user of the protocol should attempt to identify the source of the information. As specified in the draft, a source MUST forward BYE notification messages at least 5 times from each SSRC once it receives a report. It MUST NOT adjust the group size estimator or receiver RTCP bandwidth fields until the BYE SSRC has been announced at least 5 times, providing receivers with enough time detect the compromise and initiate the SSRC collision detection algorithm. In the extreme, large numbers of false BYE reports sent to a source might cause instability in a group with respect to fluctuation of joins and leaves, however the calculated group reporting interval should not be adversely affected. 11.6 Discussion of trust models As identified in the previous sections, source authenticity is a fundamental requirement of the protocol, however it is important to also clarify the model of trust that would be acceptable to achieve this requirement. There are two fundamental models that apply in this instance: a) The shared key model where all authorised group members have the same key and can equally encrypt/decrypt the data. This method assumes that an out-of-band method is applied to the distribution of the shared group key ensuring that every key-holder is individually authorized to receive the key, and in the event of Chesterfield, et al. Internet Draft - Expires December 2002 [Page 24] RTCP with Unicast Feedback member departures from the group, a re-keying exercise can occur. The advantage of this model is that the costly processing associated with one-way key authentication techniques is avoided, as well as the need to execute additional cipher operations with alternative key sets on the same data set, e.g. in the event that data confidentiality is also applied. The disadvantage is that for very large groups where the receiver set becomes effectively untrusted, a shared key does not offer much protection. b) The public-key authentication model using cryptosystems such as RSA-based or PGP authentication provides a more secure method of source authentication at the expense of generating higher processing overhead. This is typically not recommended for Real- time data streams, but in the case of RTCP reports which are distributed with a minimum interval of 5 seconds, this is potentially an option (the processing overhead might still be too great for small, low-powered devices and should therefore be considered with caution). Wherever possible, however the use of public source authentication is preferable to the shared key model identified above. As concerns requirements for protocol acceptability, either model is acceptable, although it is RECOMMENDED that the more secure public- key based options should be applied wherever possible. 11.7 Discussion of suitable security solutions This section presents some existing security mechanisms that would be suitable for addressing the requirements outlined in section 11.5. This is only intended as a guideline and it is acknowledged that there are many other solutions that would adequately address the requirements. Initial distribution of the SDP parameters for the session should use a secure mechanism such as the SAP authentication framework which allows an authentication certificate to be attached to the session announcements. Other methods might involve HTTPS or signed email content from a trusted source, however some commonly used techniques for distributing session initiation information and starting media streams are RTSP [21] and SIP [22]. RTSP provides a client or server initiated stream control mechanism for real-time multimedia streams. The session parameters are conveyed using SDP syntax and may adopt standard Transport Layer Security techniques such as are common in HTTP transactions. In other words, in order to securely retrieve and authenticate the source of the session description information such as the multicast session address and the unicast feedback identifier, the RTSP client and server should use a transport level security transaction, e.g. SSL incorporating X.509 style certificates. A typical use of SIP involving a unicast feedback identifier might be a client wishing to dynamically join a multi-party call on a multicast address using unicast RTCP feedback. The client would be Chesterfield, et al. Internet Draft - Expires December 2002 [Page 25] RTCP with Unicast Feedback required to authenticate the SDP session descriptor information returned by the SIP server. The recommended method for this, as outlined in the SIP specification [22] is to use an S/MIME message body containing the session parameters signed with an acceptable certificate. Assuming some prior knowledge or established chain of trust, the mechanisms of which are beyond the scope of this document, this would provide the client with satisfactory knowledge of the authenticity and integrity of the session descriptor information. Other options exist within the SIP specification for establishing authenticity of data such as end-to-end SIP message tunneling. For the purposes of this profile, it is acceptable to use any suitably secure authentication mechanism which establishes the identity and integrity of the information provided to the client. SRTP is the recommended AVT security framework for RTP sessions. It specifies the general packet formats and cipher operations that are used, and provides the flexibility to select different stream ciphers based on preference/requirements. It can provide confidentiality of the RTP and RTCP packets as well as protection against integrity compromise and replay attacks. It provides authentication of the data stream using the shared key trust model. Any suitable key-distribution mechanism can be used in parallel to the SRTP streams, however MIKEY [18] is the preferred system by the authors. A more general group security profile which might be used is the Group Domain of Interpretation [19], which defines the process of applying IPSec mechanisms to multicast groups. This requires the use of ESP in tunnel mode as the framework and it provides the capability to authenticate either just using a shared key or on an individual basis. It should be noted that using IPSec would break the 'transport independent' condition of RTP and would therefore not be useable for anything other than IP based communication. TESLA [20] is a scheme that provides a more flexible approach to data authentication using time-based key disclosure. The authentication uses one-way pseudo-random key functions based on key chain hashes that have a short period of authenticity based on the key disclosure intervals from the source. As long as the receiver can ensure that the encrypted packet is received prior to the key disclosure by the source, this requires loose time synchronization between source and receivers, it can prove the authenticity of the packet. The scheme does introduce a delay into the packet distribution/decryption phase due to the key disclosure delay however the processing overhead is much less than other standard public-key mechanisms. 12. Backwards Compatibility The use of unicast feedback to the source should not present any serious backwards compatibility issues. The RTP data streams should remain unaffected, as are the RTCP packets from the source enabling inter-stream synchronization in the case of multiple streams. The Chesterfield, et al. Internet Draft - Expires December 2002 [Page 26] RTCP with Unicast Feedback unicast transmission of RTCP data to a source that does not have the ability to reflect traffic by either mechanism could have serious security implications as outlined in Section 11, but would not actually break the operation of RTP. For RTP compliant receivers that do not understand the unicast mechanism, the RTCP traffic may still reach the group, in the event that an ASM distribution network is used, in which case there may be some duplication of traffic due to the reflection channel, but this should be ignored. It is anticipated, however that typically the distribution network will not enable the receiver to multicast RTCP traffic, in which case the data will be lost, and the RTCP calculations will not include those receivers. It is RECOMMENDED that any session that may involve non- unicast capable clients should always use the simple packet reflection mechanism to ensure that the packets received can be understood by all clients. 13. IANA Considerations The following contact information shall be used for all registrations included here: Contact: Joerg Ott mailto:jo@acm.org tel:+49-421-201-7028 Based on the guidelines suggested in [10], this document proposes 1 new RTCP packet format to be registered with the RTCP Control Packet type (PT) Registry: Name: RSI Long name: Receiver Summary Information Value: 208 Reference: This document. This document defines a substructure for RTCP RSI packets. A new sub-registry needs to be set up for the report block type (RBT) values for the RSI packet, with the following registrations created initially: Name: IPv4 Address Long name: IPv4 Feedback Target Address Value: 0 Reference: This document. Name: IPv6 Address Long name: IPv6 Feedback Target Address Value: 1 Reference: This document. Chesterfield, et al. Internet Draft - Expires December 2002 [Page 27] RTCP with Unicast Feedback Name: DNS Name Long name: DNS Name indicating Feedback Target Address Value: 2 Reference: This document. Name: Jitter Long name: Jitter Distribution Value: 4 Reference: This document. Name: RTT Long name: Round-trip time distribution Value: 5 Reference: This document. Name: Cumulative loss Long name: Cumulative loss distribution Value: 6 Reference: This document. Name: Loss Long name: Cumulative loss distribution Value: 7 Reference: This document. Name: Collisions Long name: SSRC Collision list Value: 8 Reference: This document. Name: BYE Long name: BYE list Value: 9 Reference: This document. Name: Stats Long name: General statistics Value: 10 Reference: This document. Name: RTCP BW Long name: RTCP Bandwidth indication Value: 11 Reference: This document. The value 3 shall be reserved for a further way of specifying a feedback target address. The value 3 MUST only be allocated for a use defined in an IETF Standards Track document. Further values may be registered on a first-come first-serve basis. For each new registration, it is mandatory that a permanent, stable, and publicly accessible document exists that specifies the semantics of the registered parameter as well as the Chesterfield, et al. Internet Draft - Expires December 2002 [Page 28] RTCP with Unicast Feedback syntax and semantics of the associated sub-block. The general registration procedures of [16] apply. 14. Outstanding Issues a) The source currently transmits RSI packets only with its own bit rate budget. This means that the receiver share of the RTCP bandwidth is unused and the granularity for information conveyed to the receivers (particularly for the group size) is inherently limited. Shall we allow, optionally, for extra bandwidth from the distribution source to the receivers to convey RSI and other information more frequently? How about the five second rule? b) We may need/want to explicitly indicate the part of the group covered by each RSI report along with further information describing the sampling interval. c) Currently, BYE announcements are repeated several times in the RSI report which is not really necessary (but follows the same behavior used for SSRC collision detection). Should we keep those two aligned or make BYE simple to handle. 15. References [1] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP - A Transport Protocol for Real-time Applications," Internet Draft, draft-ietf-avt-rtp-new-12.txt, Work in Progress, March 2003. [2] Pusateri, T, "Distance Vector Multicast Routing Protocol", Internet Draft draft-ietf-idmr-dvmrp-v3-10.txt, Work in Progress, August 2000 [3] B. Fenner, M. Handley, H. Holbrook, I. Kouvelas, "Protocol Independent Multicast - Sparse Mode (PIM-SM): Protocol Specification (Revised)", Internet Draft, draft-ietf-pim-sm-v2-new-02.txt, Work in Progress, March 2001. [4] Farinacci, D, Kouvelas, I, Windisch, K, "State Refresh in PIM- DM", Internet Draft, draft-ietf-pim-refresh-02.txt, Work in Progress, November, 2000. [5] Thaler, D, Cain, B, "BGP Attributes for Multicast Tree Construction", Internet Draft draft-ietf-idmr-bgp-mcast-attr-00.txt, Work in Progress, February 1999. [6] D. Meyer, B. Fenner, "Multicast Source Discovery Protocol (MSDP)", Internet Draft draft-ietf-msdp-spec-14.txt, Work in Progress, November 2002. [7] D. Meyer, R. Rockell, G. Shepherd, "Source-Specific Protocol Independent Multicast in 232/8", Internet Draft draft-ietf-mboned- ssm232-04.txt, Work in Progress, January 2003. Chesterfield, et al. Internet Draft - Expires December 2002 [Page 29] RTCP with Unicast Feedback [8] H. Holbrook, B. Cain, B. Haberman, "Using IGMPv3 For Source- Specific Multicast", Internet Draft draft-holbrook-idmr-igmpv3-ssm- 03.txt, Work in Progress, November 2002. [9] Session Directory Rendez-vous (SDR), developed at University College London by Mark Handley and the Multimedia Research Group, http://www-mice.cs.ucl.ac.uk/multimedia/software/sdr/. [10] Alvestrand, H. and T. Narten, "Guidelines for Writing an IANA Considerations Section in RFCs", BCP 26, RFC 2434, October 1998. [11] M. Handley, C. Perkins, E. Whelan, "Session Announcement Protocol (SAP)", RFC 2974, October 2000. [12] A. Frier, P. Karlton, and P. Kocher, "The SSL 3.0 Protocol", Netscape Communications Corp., Nov 18, 1996. [13] A. Perrig, R. Canetti, B. Whillock, "TESLA: Multicast Source Authentication Transform", Internet Draft draft-ietf-msec-tesla- spec-01.txt, Work in Progress, October 2002. [14] M. Baugher, D. McGrew, D. Oran, R. Blom, E. Carrara, M. Naslund, and K. Norrman, "The Secure Real Time Transport Protocol", Internet Draft draft-ietf-avt-srtp-05.txt, Work in Progress, June 2002. [15] B. Quinn, R. Finlayson, "SDP Source-Filters", Internet Draft draft-ietf-mmusic-sdp-srcfilter-05.txt, Work in Progress, May 2003. [16] M. Handley, V. Jacobson, and C. Perkins, "SDP: Session Description Protocol", Internet Draft draft-ietf-mmusic-sdp-new-11, Work in Progress ,November 2002. [17] T. Friedman, R. Caceres, and A. Clark (eds), "RTCP Reporting Extensions", Internet Draft, draft-ietf-avt-rtcp-report-extns- 02.txt, Work in Progress, February 2003. [18] J. Arkko, E. Carrara, F. Lindholm, M. Naslund, and K. Norrman, "MIKEY: Multimedia Internet Keying", Internet Draft draft-ietf-msec- mikey-06.txt, Work in Progress, February 2003. [19] M. Baugher, T. Hardjono, H. Harney, and B. Weis, "The Group Domain of Interpretation", Internet Draft draft-ietf-msec-gdoi- 07.txt, Work in Progress, December 2003. [20] A. Perrig, R. Canetti, B. Whillock, "TELSA: Multicast Source Authentication Transform Specification", Internet Draft draft-ietf- msec-tesla-spec-00.txt, Work in Progress, October 2002. [21] H. Schulzrinne, A. Rao, R. Lanphier, "Real Time Streaming Protocol (RTSP)", RFC 2326, April 1998. Chesterfield, et al. Internet Draft - Expires December 2002 [Page 30] RTCP with Unicast Feedback [22] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, June 2002. 16. Appendix A Distribution Report processing at the receiver A.1 Algorithm Example processing of Loss Distribution Values X values represent the loss percentage. Y values represent the number of receivers. Number of x values is the NDB value xrange = Max Distribution Value(MaDV) - Min Distribution Value(MnDV) First data point = MnDV,first ydata then Foreach ydata => xdata += (MnDV + (xrange / NDB)) A.2 Pseudo-code Packet Variables -> factor,NDB,MnDVL,MaDV Code variables -> xrange, ydata[NDB],x,y xrange = MaDV - MnDV x = MnDV; for(i=0;i 20 Bytes) Minimum Data Value: 0 Maximum Data Value: 39 Data Bucket values: 13029, 352, 5460, 855 (each value is 16-bits) Results, 16-bit buckets: X - 0 - 9 | 10 - 19 | 20 - 29 | 30 - 39 Y - 13029 | 352 | 5460 | 855 Example - 2nd Method Description A semi-accurate method for representing data. This method wishes to optimise the space used to convey results while representing every value. Algorithm Attempt to convey all values, i.e. 40 buckets, but in the smallest amount of space possible by using the multiplicative factor. The maximum value is 3120. Using the maximum multiplicative factor this will not fit into a 2, 4 or 6 bit space. The next minimum size block is 8 bits. The maximum value of 3120 will fit into an 8 bit space using the lowest possible multiplicative factor of 13. Can the whole of the data set fit into a maximum size packet? The maximum packet size is 1008 bytes. This will fit since there are only 39 data points. The packet fields will contain the values: Chesterfield, et al. Internet Draft - Expires December 2002 [Page 34] RTCP with Unicast Feedback Header Distribution Block Distribution Type: 1 Number of Data Buckets: 40 Multiplicative Factor: 13 Packet Length field: 13 (13 * 4 => 52 Bytes) Minimum Loss Value: 0 Maximum Loss Value: 39 Data Portion Results, 8-bit buckets: X - 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | 9 Y - 77 | 62 | 0 | 138 | 200 | 240 | 177 | 85 | 15 | 8 X - 10 | 11 | 12 | 13 | 14 | 15 | 16 | 17 | 18 | 19 Y - 6 | 2 | 2 | 5 | 5 | 6 | 0 | 1 | 0 | 0 X - 20 | 21 | 22 | 23 | 24 | 25 | 26 | 27 | 28 | 29 Y - 0 | 1 | 7 | 177 | 89 | 21 | 18 | 16 | 15 | 16 X - 30 | 31 | 32 | 33 | 34 | 35 | 36 | 37 | 38 | 39 Y - 13 | 13 | 8 | 7 | 6 | 4 | 5 | 6 | 3 | 0 Example - 3rd Method Description This demonstrates the most accurate method for representing the data set. This method doesn't attempt to optimise any values. Algorithm Identify the highest value and select buckets large enough to convey the exact values, i.e. no multiplicative factor. The highest value is 3120. This requires 12 bits (closest 2 bit boundary) to represent, therefore it will use 60 bytes to represent the entire distribution. This is within the max packet size, therefore all data will fit within one report block. The multiplicative value will be 1. The packet fields will contain the values: Header Distribution Block Distribution Type: 1 Number of Data Buckets: 40 Multiplicative Factor: 1 Packet Length field: 18 (18 * 4 => 72 Bytes) Minimum Loss Value: 0 Maximum Loss Value: 39 Bucket values are the same as initial data set. Result Chesterfield, et al. Internet Draft - Expires December 2002 [Page 35] RTCP with Unicast Feedback The selection of which of the three methods outlined above might be determined by a congestion parameter, or a user preference. The overhead associated with processing the packets is likely to differ very little between the techniques. The savings in bandwidth are apparent however, using 20, 52 and 72 octets respectively. These values would vary more widely for a larger data set with less correlation between results. C AUTHORS ADDRESSES Julian Chesterfield University of Cambridge Computer Laboratory, JJ Thompson Avenue, Cambridge, CB3 0FD, UK Julian.chesterfield@cl.cam.ac.uk Eve Schooler AT&T Labs - Research 75 Willow Road Menlo Park, CA 94025 eve_schooler@acm.org Joerg Ott Tellique Kommunikationstechnik GmbH Berliner Str. 26 D-13507 Berlin GERMANY Phone: +49.30.43095-560 (sip:jo@tzi.org) Fax: +49.30.43095-579 Email: jo@tellique.com D FULL COPYRIGHT STATEMENT Copyright (C) The Internet Society (2002). All Rights Reserved. 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