Presentation on theme: "Multipath RTP Applying Multipath Communication to Real time Applications By: Saba Ahsan Supervisor: Prof. Jörg Ott Conducted at Comnet, Aalto University."— Presentation transcript:
Multipath RTP Applying Multipath Communication to Real time Applications By: Saba Ahsan Supervisor: Prof. Jörg Ott Conducted at Comnet, Aalto University of Science & Technology
Motivation Most transport protocols select a single path for communication flow between two end hosts, even when multiple paths exist. Such flows are unable to fully utilize the available resources. Multihomed clients have more than one network interface. Multipath capability refers to the simultaneous use of multiple paths through the network, which may significantly improve performance and reliability. In real-time communication this could would improve the end-user experience by enhancing the QoS. – Bandwidth-hungry applications such as video streaming and IP-TV can benefit from the increased, combined throughput available to multihomed clients – retransmission of lost data is often uncharacteristic of real-time traffic because of time constraints; multipath senders can avoid lossy paths or send redundant data over multiple paths – session-based real-time communication can benefit from the redundancy by implementing failover in case of network failures
Problem Statement Design a solution for the transport of real-time data while simultaneously using multiple paths on multihomed clients. Design of Multipath RTP (MPRTP) an extension of RTP protocol, with multipath capabilities. Design, implementation and testing of an MPRTP based solution for video streaming to a single receiver, when multiple paths exist between the sender and receiver.
Background Real-Time Protocol (RTP) RTP is an end-to-end protocol designed for transporting real-time traffic such as voice and video, over multicast and unicast. It uses RTCP (Real-time Control Protocol) for monitoring the transmission quality, however it does not guarantee QoS. It is independent of the transport and network layers. A signalling protocol such as SIP or SDP is used for managing RTP sessions. It does not care about congestion control or fair usage like TCP.
MPRTP Protocol Internet Draft : https://datatracker.ietf.org/doc/draft-singh-avt- mprtp/https://datatracker.ietf.org/doc/draft-singh-avt- mprtp/
MPRTP Goals Increased Throughput Concurrent use of paths such that the combined available capacity is higher than the capacity of any individual path Improved Reliability MPRTP should be able to transmit redundant streams on different paths for reliability and support fallback in case of path failures for robustness. Compatibility – Application Compatibility: MPRTP stack must be capable of working with legacy RTP applications. – Network Compatibility: MPRTP subflows should appear as RTP flows and be able to traverse through NATs and Firewalls.
Architecture Application MPRTP RTP … UDP/TCP … IP … Physical … Each path represents an MPRTP subflow. Like RTP, MPRTP can work with different transport protocols.
MPRTP Specification Path Management Path awareness and management of port+IP pair bindings. MPRTP is designed to use in-band signaling for path advertisements and/or connectivity checks. Interface discovery may be done using ICE. Packet Scheduling Splitting of data into multiple subflows across different paths. Subflow recombination Recombining the subflows, so that it appears as a single stream to the application MPRTP Sender MPRTP Receiver Internet subflow 1 subflow 2 subflow 3 subflow 2 subflow 3 MPRTP Flow Gather characteristics of different paths and schedule packets accordingly Reorder the data correctly and hand over to application, send reports about quality
MPRTP Header Extension for RTP RTP sequence numbers used for packet reordering of the stream Flow-specific sequence numbers increase monotonically for each path, independent of other paths. 0123456789 1010123456789 2020123456789 30301 V=2PXCCMPTSequence number Timestamp Synchronization source (SSRC) identifier Contributing source (CSRC) identifiers …… RTP H-Ext IDlengthMPR_Type Flow IDFlow specific sequence number RTP payload …….
MPRTP Implementation RAMP-UP RTP Adaptation for Multiple Paths – Using Percentage distribution
Ordering is based on overall sequence number. Packets are inserted into the jitter buffer as soon as they arrive. Playout starts after a predefined latency period. We use 2 seconds for our testing. Late packets are discarded 95864 07321 Time Path 1 Path 2 Reordering in Jitter Buffer 9586407321 RAMP-UP Receiver
RAMP-UP sender is designed for video streaming across multiple paths. We assume that the bitrate of the video is higher than the bitrate of any of the available paths and hence it is necessary to use more than one path. A common bottleneck may exist. However, if it is common to all available paths, then congestion/losses can not be avoided. The sender is not capable of reducing video bitrate. We utilize a non-aggressive approach, which implies that we do not put more traffic on a path unless necessary. This in turn implies that the full capacity of certain paths may never be known. RAMP-UP Sender (1)
RAMP-UP uses percentage distribution on the paths. Percentages are assigned according to measured characteristics. This approach eliminates flapping and ensures equal distribution of traffic on all paths if video rate is increased. Measurements are based on data gathered by RTCP reports. All packets of a frame are sent on the same path. Initially, equal percentage is assigned to each path. RAMP-UP Sender (2) Path 1 Queue To Receiver Sender’s Buffer Decision based on percentage level Path 2 Queue
Using RTCP reports, the sender is able to calculate the bitrate observed on each path (TB i ) The observed bitrate depends on the amount of traffic being sent on the path, hence the sender would only update bitrate values if they are higher than what was previously recorded, except if the ratio of lost packets (L i ) is greater than 0 in the RTCP RR. Average packet size (S i ) is calculated for each interval during which the bitrate was measured. If a path has continuous losses, it is considered congested, and in this case the observed bitrate is stored in another variable called CB i. The sender assigns a percentage of traffic to each path using TB i values if path is not congested, and CB i value if path is congested. Congestion condition is cleared if losses don’t appear for a predefined amount of time ( in our testing we use 25 seconds) RAMP-UP Sender (3)
RAMP-UP Sender (4) HSN = 1000 HSN 2 = 2300, t 2 HSN 1 = 1000, t 1 t 2 -t 1 HSN = 2300 SenderReceiver RTP RTCP
Test Setup The test environment consists of virtual machines running on a single physical machine. Network properties are emulated using Network Emulator (NetEm) MPRTP Sender MPRTP Receiver Router1 Router2 Router3 Virtual Environment
Results: Bitrate of data being sent on each path as measured by the sender when three paths are available. Path capacities are changed during simulation. Time (s) Bitrate (kpbs)
Results: Percentages assigned to the paths over time Total Packets lost = 1.6%, Frames lost = 6%, BER = 1.8% Time (s) Assigned Ratios
Conclusion It is possible to achieve higher bitrates using multiple paths, which may help streaming higher quality videos. The extra paths may be used to avoid losses due to temporary congestion on any of the paths. Video streaming, is just one of the many applications of MPRTP. MPRTP protocol is still in its infancy. RAMP-UP focuses on video streaming only. The scheduling algorithm can be improved further. Multiple streams (voice/video, lip-sync) and rate-control mechanism can be incorporated. Many research opportunities arise from this study – MPRTP for mobile environments, 3G, GPRS & WLAN interfaces – MPRTP for voice fallback