Internet Engineering Task Force AVT WG Internet Draft Julian Chesterfield draft-chesterfield-avt-rtcpssm-02.txt AT&T Internet Research Joerg Ott Tellique Kommunikationstechnik GmbH November, 2001 Expires: May, 2002 RTCP Extension for Single Source Multicast Sessions with Unicast RTCP 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 to enable the operation of RTP/RTCP using unicast RTCP feedback for Single Source multicast sessions such as Source Specific Multicast (SSM) Communication where the traditional model of Any Source Multicast (ASM) group communication of many to many is either not possible or not preferred. This draft can be applied to any group communication which might benefit from a sender controlled summarised reporting mechanism. It extends [1], section 6 which defines the RTP session group control channel. 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. Chesterfield, Ott [Page 1] Internet Draft RTCP with Unicast Feedback November, 2001 2. Introduction RTP [1] provides a real-time transport mechanism suitable for unicast or Internet Standard Multicast communication between multimedia applications. Typical uses are for real-time or near real-time group communication via 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 the distribution or calculation of session specific information such as packet loss, and round trip time calculation 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 and establish network conditions and diagnose faults based on receiver locations. RTP was designed to operate in a unicast mode or in the traditional mode of Any Source Multicast (ASM) Group communication which encompasses a network which supports both one to many and many to many communication via a common group address in the range 224.0.0.0 through 239.255.255.255. Typical routing protocols that enable such communication are Distance Vector Multicast Routing Protocol (DVMRP) [2] or Protocol Independent Multicast (PIM) [3][4] Sparse/Dense Mode in combination with an Inter-domain routing protocol such as Multicast Border Gateway Protocol (MBGP) [5] with Multicast Source Discovery (MSDP) [6]. Such routing protocols enable a host to join a single Multicast group address and send to or receive traffic from all members in the group with no prior knowledge of membership. In order to enable such a service in the network, however there is a great deal of complexity involved at the routing level. The Source Specific Multicast (SSM) [7] Model has the advantage of removing a great deal of the routing complexity involved in multicast group creation and source information distribution. The disadvantage of SSM with respect to Real-time traffic using RTP is that the simplification to the routing protocols removes the ability for any member of the group to communicate with any other member of the group without an explicit SSM (Source, Group) or unicast join to that host. The solution proposed in this draft defines a new method for distributing control information amongst all members in a multicast session and is designed to operate under any multicast group communication scenario. It is, however, of particular benefit to SSM sessions in the absence of receivers being able to communicate with each other. The RTP data stream protocol itself is unaffected. The basic architectural models to which this feedback method could apply are: a) SSM groups with a single sender. This is the main motivation behind the unicast RTCP feedback mechanism and allows SSM groups which do not have many to many communication capability, traditionally available in ASM multicast groups to still receive RTP data streams and operate on them in the usual manner. SSM adopts the notion of a sender data channel which provides a one to many communication facility from the source to all the receivers in the Chesterfield, Ott [Page 2] Internet Draft RTCP with Unicast Feedback November, 2001 group. The feedback is unicast to the source on the standard RTCP port. b) One to many broadcast networks such as satellite communication typically using a terrestrial link low bandwidth return channel or a broadband cable link. This architecture differs very little from the SSM channel concept, but most likely will require a translator of some kind to render the RTP data stream onto the satellite or cable distribution channel. c) ASM with a single sender. An SDP session announcement type identifies a session as having a single sender receiving unicast RTCP feedback. Receivers join the multicast group address and receive RTP and RTCP data on the specified address/port combinations. The RTCP feedback is directed to the source on the RTCP port. This model is not considered to be more efficient than a standard multicast group RTP communication scenario, and is therefore not recommended to replace the traditional mechanism, however it might be useful in helping to prevent overtaxing multicast routing infrastructure that does not scale as efficiently. SSM sessions are typically assigned a value in the group address range 232.0.0.0 through 232.255.255.255, although this is not a requirement. A session may be assigned any valid multicast address, as long as the local network is configured to allow source specific joins outside the suggested SSM range. In order for a host to receive traffic from an SSM capable source, it must support the IGMPv3 multicast group membership reporting protocol which enables the host to explicitly request traffic from a source,group pair. In this case, the host is aware of the significance of the address range and is therefore capable of identifying the unicast RTCP feedback session requirements based on this knowledge. For sessions which take advantage of the unicast feedback model but do not inherently need to use it, it is anticipated that an SDP syntax will be defined. The modifications proposed in this document are intended to provide an optional replacement to the method of RTCP operation for sessions which either require or may benefit from a new reporting structure. For certain distribution networks, such as SSM networks, this may be a requirement, however in other cases this is an optional feature which may be used. 3. Basic Operation This draft proposes two methods for enabling receiver feedback to all members in a session. Each involves the unicasting of RTCP packets to a source whose job it is to distribute the information to the members of the group. The source must always be able to communicate with all the other members in order for either mechanism to work. The first method, the 'Simple Feedback Model' is a basic mechanism whereby all receiver reports are unicast to the source and Chesterfield, Ott [Page 3] Internet Draft RTCP with Unicast Feedback November, 2001 subsequently forwarded by the source to all receivers on the multicast feedback channel. The advantage of using this method is that an existing receiver implementation requires little modification in order to operate in this new state. Instead of forwarding Receiver Reports to a multicast address, it uses a unicast address and still receives RTCP traffic in the usual manner. This method also has the advantage of being backwards compatible with RTP/RTCP implementations which do not support unicast feedback to the source and operate using the standard multicast group communication model, ASM. In a session that is using ASM, such a receiver would multicast Receiver Reports to the group address and port+1 as stated in [1]. This would still be received by all receivers. In a session using an SSM distribution network, the network would prevent any data from the receiver being distributed further than the first hop router. Additionally, any data heard from this receiver by other hosts on the same subnet should be filtered out by the host IP stack and will therefore not cause any problems with respect to the calculation of Receiver RTCP bandwidth since this receiver will not be heard by any other members. The second method, the 'Sender Feedback Summary Model' is a summarised reporting scheme that provides savings in bandwidth by consolidating all the receiver reports into one summary packet which is then distributed to all the receivers. The advantage of this scheme is apparent for large group sessions where the basic forwarding mechanism outlined above would create a large amount of packet replication in order to forward all the information to all the receivers. The basic operation of the scheme is the same as the first method, however it requires that all the members in the session understand the new summarised packet format outlined in section 7.1. To differentiate between the two reporting mechanisms, a new SDP identifier is created and discussed in section 10. The method of reporting must be decided prior to the start of the session, a distribution source may not change the method during a session. 4. Definitions Distribution Source: In order for unicast feedback to work, there must only be one session distribution source for any subset of receivers to which RTCP feedback is directed. 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 (see section 9 for details on this) to create a single source group. However, in order for unicast feedback to work, only one source must be responsible for distributing the RTP stream and forwarding RTCP information to all receivers. RTP and RTCP Channels: The data distribution from the source to the receivers whether via an SSM {source,group} identifier, a standard ASM multicast group or a unicast reflector, is referred to as the RTP and RTCP channels. These channels are differentiated via the port numbers as [port] and [port + 1] for RTP and RTCP respectively. See [1] for further explanation of the port numbering. Chesterfield, Ott [Page 4] Internet Draft RTCP with Unicast Feedback November, 2001 Unicast RTCP Feedback Target: For a session defined as having a distribution source A, on ports n and n+1, the unicast feedback target is the IP address of Source A on port n+1. SSRC: Synchronization source. A 32-bit value that uniquely identifies each member in a session. See [1] for further information. Report blocks: In the RTCP design [1] it is encouraged to stack multiple report blocks in Sender and Receiver report packets. In this way, a variable size packet is created which can include information from one source pertaining to multiple sources in the group. The concept of report blocks is extended in this draft to encompass Loss Jitter Summary packets in which a source can optionally stack multiple reports into one packet in order to provide additional feedback on the RTCP traffic received from the group. 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 These remain unmodified. In addition to the exisiting types, two new packet types are introduced. Further information on each of these is provided in this draft. The packet types are: type description Payload number RSI Receiver Summary Information [see section 12] LJS Loss and Jitter Summary [see section 12] 6. Simple feedback model 6.1 Packet Formats For this mechanism, the packet types used remain the same as for standard RTCP feedback in [1]. Receivers generate Receiver Reports with information on the quality of the stream received from the source. The source must create Sender Reports which 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 usual rules for BYE and APP packets also apply. 6.2 Distribution Source behaviour For the simple feedback model, the source provides a simple packet Chesterfield, Ott [Page 5] Internet Draft RTCP with Unicast Feedback November, 2001 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. The source may 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 to the group on the multicast RTP channel. Any multicast data received on this port 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 it's own control data bandwidth allowance. The algorithm for computing the allowance is explained in section 9. The control bandwidth traffic included in the calculation includes any Sender reports to the group, along with any additional SDES and APP packets. 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. This decision must be made in advance of the session and indicated in the SDP announcement. 6.3 Receiver behaviour Receivers listen on the RTP and RTCP channels for data. Each receiver calculates it's share of the receiver bandwidth based on the standard rules i.e. 75% of the RTCP bandwidth is divided equally between all unique SSRCs in the session. See section 9 for further information on this. 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 mode, the sender is required to summarise the information received from all the Receiver Reports generated by the receivers and place the information into summary reports. The sender must send at least 1 Receiver Summary Information packet for each reporting interval. The sender can additionally stack Loss Jitter Summary (LJS) reports after the RSI packet. Each LJS packet 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 LJS packets are not required but recommended. RSI and LJS packets are sent in addition to the standard Sender Reports and SDES packets outlined in [1]. 7.1 Packet Formats The Sender feedback summary model introduces 2 new packet formats. The Receiver Summary Information packet (RSI) which must be sent by a source if the summarised feedback mechanism is selected and the optional Loss and Jitter Summary report packet (LJS) that may be Chesterfield, Ott [Page 6] Internet Draft RTCP with Unicast Feedback November, 2001 appended to the RSI packet to provide more detailed information on the overall session characteristics reported by all receivers. 7.1.1 RSI: Receiver Summary Information RTCP Packet 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| SC | PT | length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of Sender | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | group size | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | AFL | HCNL | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Highest interarrival jitter | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Receiver RTCP Bandwidth | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | collision SSRC #1 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | . . . | The RSI packet consists of a main report block modeled along the same lines as a receiver report with optional LJS blocks appended. The first 4 bytes of header extension follow the standard RTP header outline. This ensures backwards compatibility with older versions which may not understand the RSI packet format but can read the length field indicating the end of the report block. The following fields are included: The fields "V", "P", and "length" have the same meaning as per [1]. SC: 5 bits The number of collision SSRC entries towards the end of the report block. A value of 0 is allowed. SSRC: 32 bits The synchronisation source identifier for the originator of the summary report packet. group size: 32 bits This field provides the sender's view of the number of receivers in a session. This should 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]. 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 point at the left edge of the field. Chesterfield, Ott [Page 7] Internet Draft RTCP with Unicast Feedback November, 2001 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. Highest interarrival jitter: 32 bits Highest 'interarrival jitter' value out of all RTCP RR packets since the last RSI from any of the receivers. receiver 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. collision SSRC: n x 32 bits 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 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 and it is the responsibility of the receiver(s) to detect the collision and select another ID. 7.1.2 LJS: Loss Jitter Summary RTCP Packet 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| LJSC | PT | Length | header +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of Sender | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | NLB | LF | MIL | MAL | NJB | JF | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Minimum Jitter Value | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Maximum Jitter Value | report +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block | Loss Buckets | 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Loss Buckets cont... | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Jitter Buckets | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Jitter Buckets cont... | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | ... | report | ... | block +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ 2 Loss Jitter Summary Count (LJSC): 5 bits The number of Loss Jitter Summary report blocks contained in this packet Chesterfield, Ott [Page 8] Internet Draft RTCP with Unicast Feedback November, 2001 Number of Loss Buckets (NLB): 4 bits The number of Loss Buckets over the reserved 64 bit space. Possible values are 0, 1, 2, 4, 8, 16 Loss Factor (LF): 4 bits Indicates the multiplicative factor to be applied to the Loss Bucket values. Possible values are 1 - 15. Minimum Loss Value (MIL): 8 bits Minimum loss value. In combination with the Maximum Loss value indicates the range covered by the Loss Bucket values. Possible values are 0 - 99. The Minimum Loss Value must always be less than the maximum, expressed as a fixed point number with the binary point at the left edge of the field. Maximum Loss Value (MAL): 8 bits Maximum loss value. In combination with the Minimum Loss value indicates the range covered by the Loss Bucket values. Possible values are 1 - 100. The maximum Loss Value must always be greater than the minimum, expressed as a fixed point number with the binary point at the left edge of the field. Number of Jitter Buckets (NJB): 4 bits The number of Jitter Buckets over the reserved 64 bit space. Possible values are 0, 1, 2, 4, 8, 16 Jitter Factor (JF): 4 bits Indicates the multiplicative factor to be applied to the Jitter Bucket values. Possible values are 1 - 15. Minimum Jitter Value (MIJ): 32 bits Minimum jitter value. In combination with the Maximum jitter value indicates the range covered by the jitter Bucket values. The Minimum jitter Value must always be less than the maximum. Maximum Jitter Value (MAJ): 32 bits Maximum jitter value. In combination with the Minimum jitter value indicates the range covered by the jitter Bucket values. The Maximum jitter Value must always be greater than the minimum. Loss Buckets: 16*4 bits - 8*8 bits - 4*16 bits - 2*32 bits - 1*64 bits Loss Bucket. The size and number of buckets depends upon the value of NLB. This indicates the division of the 64 bit space. Depending upon whether NLB is 16, 8, 4, 2 or 1, the size of each LB will be 4, 8, 16, 32 or 64 bits respectively. Each value must be multiplied by the Loss Factor. Jitter Buckets: 16*4 bits - 8*8 bits - 4*16 bits - 2*32 bits - 1*64 bits Jitter Bucket. The size of the bucket depends upon the value of JLB. This indicates the division of the 64 bit space. Depending upon whether JLB is 16, 8, 4, 2 or 1, the size of each JB will be 4, 8, 16, 32 or 64 bits respectively. Each value must be multiplied by the Jitter Factor. Chesterfield, Ott [Page 9] Internet Draft RTCP with Unicast Feedback November, 2001 7.2 Distribution Source behaviour The length field of the RSI packet must be calculated over the length of the whole packet, using the method defined in [1]. The group size must be included in the RSI packet. The source should also calculate the Receiver RTCP bandwidth field. Typically this value will be calculated as outlined in [1] using the group size and session bandwidth as variables. This field does however provide the source with the capability to control the amount of feedback from the receivers and can be increased or decreased based on the requirements of the source. Regardless of the value selected by the source for the 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. In order to identify SSRC collisions, the source is responsible for maintaining a record of each SSRC and the correpsonding IP address within at least one reporting interval 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 does not prevent the possibility of collisions since IP addresses may not be unique e.g. due to NAT gateways, however it greatly increases the capability to detect collisions. In the event that collisions are not detected, the effect will be an innaccurate impression of the group size on the part of the source. Since the statistical probablility that collisions will both occur and be undetectable is very low, the clients would have to randomly select the same SSRC and be located behind the same NAT gateway, this should not be a significant concern. For the LJS packet, the source must decide which are the most significant values to convey. 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 the 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 RTCP reports as LJS information. 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 loss and jitter information. 7.3 Receiver behaviour The receiver must process RSI packets and adapt session parameters Chesterfield, Ott [Page 10] Internet Draft RTCP with Unicast Feedback November, 2001 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 the 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 it's share of the RTCP bandwidth. This share of the bandwidth may be overridden by the 'Receiver RTCP Bandwidth' field. This field provides the source with the capability to control the amount of feedback from the receivers. The receiver can handle the LJS 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 it's 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 for the use of mixers and translators in an RTP session 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 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 generically as a gateway. A gateway 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 gateway 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. Chesterfield, Ott [Page 11] Internet Draft RTCP with Unicast Feedback November, 2001 8.1 Use of a mixer-translator The gateway must adhere to the SDP descriptor for the single source session and use the feedback mechanism indicated. Receivers should be aware that by introducing a gateway into the session, more than one source may potentially be active in a session since the gateway 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 gateway itself. It is recommended that the simple packet reflection mechanism be used under these circumstances since attempting to coordinate RSI + LJS reporting between more than one source may be complicated unless the gateway 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 gateway must be able to follow the same security policy as the client in order to unicast forward RTCP data 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 gateway is not acting as the distribution source, and subsequent unicast feedback to the source is only allowed if the gateway can conduct the same source authentication as required by the receivers. 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 it's own SSRC count determined by the forwarded Receiver Reports, or from the RSI 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 the group. The source should calculate the transmission interval for RSI + LJS packets out of it's 1/4 of the control traffic bandwidth with a minimum transmission interval of 5 seconds. Chesterfield, Ott [Page 12] Internet Draft RTCP with Unicast Feedback November, 2001 10. SDP Extensions The Session Description Protocol (SDP) is used as a means to describe media sessions in terms of their transport addresses, codecs, and further attributes. Providing RTCP feedback via unicast as specified in this document constitutes another session parameter. To make receivers aware that they are supposed to provide their feedback via unicast, this needs to be indicated in the session description. Similarly, parameters of SSM RTCP feedback -- such as the mode of summarizing information at the sender and the target unicast address to send feedback information to -- needs to be provided. This section defines the necessary SDP parameters (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 further 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 ljs -- MUST be used to indicate the "Loss Jitter Summary" mode of operation rtcp:unicast rsi -- MUST be used to indicate the "Receiver Summary Information" mode of operation. 10.2 SSM Source Specification In addition, in an SSM RTCP session, the sender(s) need to be indicated for both source-specific joins to the multicast group as well as for addressing RTCP packets to. This is done following the proposal for SDP source filters documented in draft-ietf-mmusic-sdp-srcfilter-00.txt [15]. From this specification, only the inclusion mode ("a=incl:") MUST be used for SSM RTCP. There SHOULD be exactly one "a=incl:" attribute listing the address of the sender. The RTCP port MUST be derived from the m= line of the media description. Chesterfield, Ott [Page 13] Internet Draft RTCP with Unicast Feedback November, 2001 11. Security Considerations Packet bombing of unsuspecting victims via a fake SDP or SSM address is a real concern for this architecture. For this reason it is required that a security policy be applied to any session which involves unicast feedback of data to a single IP address. At a minimum, it is recommended that source authentication be conducted by every receiver prior to unicasting data. An additional concern is the problem of fake RSI + LJS packets which could increase the RTCP bandwidth sent to the source. Any security policy must address this as a minimum requirement. Receiver authentication would also be beneficial, since Denial of Service attacks by generating false Receiver Reports is also possible. The consequences of this are not as drastic, affecting only the group size and transmission interval calculation and therefore the integrity and frequency of the reported data. The issue of source feedback implosion should not occur in the event that receivers practice the standard RTP/RTCP guidelines for starting sessions and for implementing the scaling algorithm based on the number of hosts. An additional issue which should be addressed, but is beyond the scope of this document is the potential for host anonymity which is facilitated by Source Specific Multicast and adds additional security measures into group communication. By explicitly controlling receiver feedback, a source could solicit feedback from the receivers in a scalable way without the need to inform all members in a session of the group membership. 11.1 Security Requirements Outline In order to overcome the issues outlined above, there are some minimal and recommended policies which must be addressed: - The information providing the unicast feedback address needs to be authenticated as being from a trusted source. - Data integrity of the RTCP traffic from the source, particularly RSI + LJS packets is also required. - Receiver authentication is recommended in order to ensure integrity of RTCP traffic and group size. - Data encryption of both the RTCP and RTP streams are optional but recommended for this draft. Ideally, a public key infrastructure would provide the mechanism necessary to ensure the trusted authentication of distributed SDP announcements. Since this is not generally available, the following precautions are highly recommended. Chesterfield, Ott [Page 14] Internet Draft RTCP with Unicast Feedback November, 2001 The primary danger of the use of a fake session announcement must be addressed by the distribution media itself since SDP remains independent of the underlying mechanism and provides no facility to combat authentication and/or message integrity. The most common methods of distributing SDP messages are the Session Announcement Protocol (SAP)[11], a web page or an email message. All of these mechanisms provide the capability to authenticate the source of the announcement: - SAP has the option to include an authentication header in the message which assures the integrity of the message contents and identifies the source of the message via public key encryption. - A secure web server can be used to provide Secure Sockets Layer (SSL) [12] authentication of the web site containing the SDP message. - An email message can be signed using a public key mechanism to ensure data integrity. All of these methods rely on a level of trust in order to validate the public key of the originator of the message. The establishment of trust is beyond the scope of this document, however it is recommended that receivers should only trust an originating source if a digital certificate signed by a trusted third party such as a Signing Authority is available or if the key has been received prior to the session via some secure out-of-band method. A number of options are available to address the issue of session data integrity, the most obvious being the use of Secure RTP (SRTP) [14] or a more general security framework such as TESLA [13]. Adopting such a scheme to ensure that both source traffic and receiver messages are encrypted would prevent the generation of fake RTCP traffic to the group or from any unsolicited receivers. 12. IANA Considerations Based on the guidelines suggested in [10], this document proposes 2 new RTCP data payload types for consideration by IANA. Furthermore, four new SDP media-level attributes are defined in section 10. 13. 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-10.txt, Work in Progress, July 2001. [2] Pusateri, T, "Distance Vector Multicast Routing Protocol", draft-ietf-idmr-dvmrp-v3-10, August 2000 Chesterfield, Ott [Page 15] Internet Draft RTCP with Unicast Feedback November, 2001 [3] Fenner, B, Handley, M, Holbrook, H, Kouvelas, I, "Protocol Independent Multicast - Sparse Mode (PIM-SM): Protocol Specification (Revised)", draft-ietf-pim-sm-v2-new-02.txt, March 2001 [4] Farinacci, D, Kouvelas, I, Windisch, K, "State Refresh in PIM-DM" draft-ietf-pim-refresh-02.txt, November, 2000 [5] Thaler, D, Cain, B, "BGP Attributes for Multicast Tree Construction", draft-ietf-idmr-bgp-mcast-attr-00.txt, February 1999 [6] Farinacci, D, Rekhter, Y, Meyer, D, Lothberg, P, Kilmer, H, Hall, J, "Multicast Source Discovery Protocol (MSDP)", draft-ietf-msdp-spec-06.txt, July 2000 [7] Shepherd, G, Luczycki, E, Rockell, R, "Source-Specific Protocol Independent Multicast in 232/8", draft-shepherd-ssm232-00.txt, March 2000. [8] Holbrook, H, Cain, B, "Using IGMPv3 For Source-Specific Multicast", draft-holbrook-idmr-igmpv3-ssm-00.txt, July 2000. [9] Session Directory Rendez-vous (SDR), developed at University College London by Mark Handley and the Multimedia Research Group. [10] Alvestrand, H. and T. Narten, "Guidelines for Writing an IANA Considerations Section in RFCs", BCP 26, RFC 2434, October 1998. [11] Handley, M, Perkins, C, Whelan, E, "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] Perrig, Canetti, Briscoe, Tygar, Song, "TESLA: Multicast Source Authentication Transform", draft-irtf-smug-tesla-00.txt. [14] E. Carrara, D. McGrew, M. Naslund, K. Norrman, D. Oran, "The Secure Real Time Transport Protocol", draft-ietf-avt-srtp-01.txt. [15] B. Quinn, "SDP Source-Filters", Internet Draft draft-ietf-mmusic-sdp-srcfilter-00.txt, Work in Progress, May 2000. Chesterfield, Ott [Page 16] Internet Draft RTCP with Unicast Feedback November, 2001 13. Appendix A LJS packet processing at the receiver A.1 Algorithm X values represent the loss percentage. Y values represent the number of receivers. Number of x values is the NLB value xrange = MAL - MIL First data point = MIL,first ydata then Foreach ydata => xdata += (MIL + (xrange / NLB)) A.2 Pseudo-code Packet Variables -> factor,NLB,MIL,MAL Code variables -> xrange, ydata[NLB],x,y xrange = MAL - MIL x = MIL; for(i=0;i