Appendix A. How to Control the Number of Rtxs. per Packet
Finding out the number of retransmissions (rtxs.) per packet for achieving the best possible transmission is a difficult task. Of course, the absolute minimum should be one (1); otherwise, do not use this payload format. Moreover, as of date of publication, the authors were not aware of any studies on the number of retransmissions per packet that should be used for best performance. To help implementers and researchers on this item, this section describes an estimate of the buffering time required for achieving a given number of retransmissions. Once this time has been calculated, it can be communicated to the client via SDP parameter "rtx-time", as defined in this document.
A.1. Scenario and Assumptions
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Streaming scenario with relaxed delay bounds. Client and server are provided with buffering space as indicated by the parameter "rtx-time" in SDP.
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RTP AVPF profile used with SSRC-multiplexing retransmission scheme: 1 SSRC for original packets, 1 for retransmission packets.
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Default RTCP bandwidth share for SRs and RRs, i.e., SR+RR = 0.05. We have senders (2) and receivers (1). Receivers and senders get equally 1/3 of the RTCP bandwidth share because the proportion of senders is greater than 1/4 of session members.
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avg-rtcp-size is approximated by 120 bytes. This is a rounded-up average of 2 SRs, one for each SSRC, containing 40/8/28/32 bytes for IPv6/UDP/SR/SDES with CNAME, thus making 105 bytes each; and a RR with 40/8/64/32 bytes for IPv6/UDP/2*RR/SDES, making 157 bytes. Since senders and receivers share the RTCP bandwidth equally, then avg-rtcp-size = (157+105+105)/3 = 117.3 ~= 120 bytes. The important characteristic of this value is that it is something over 100 bytes, which seems to be a representative figure for typical configurations.
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The profile used is AVPF [1] and Generic NACKs are used for requesting retransmissions. This adds 16 bytes of overhead for 1 NACK and 4 bytes more for every additional NACK Feedback Control Information (FCI) field.
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We assume a worst-case scenario in which each packet exhausts its corresponding number of available retransmissions, N, before it is received. This means that if a packet is requested for retransmission a maximum of 2 times, the corresponding generic NACK report block requesting that particular packet is sent in two consecutive RTCP compounds; likewise, if it is requested for retransmission 10 times, then the generic NACK is sent 10 times. This assumption makes the RTCP packet size approximately constant after N*RTCP intervals (seconds), namely, to avg-rtcp-size = 120 + (receiver-RTCP-bw-share)*(12 + 4*N). In our case, the receiver RTCP bandwidth share is 1/3; thus, avg-rtcp-size = 124 + 4*N/3.
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Two delay parameters are difficult to approximate and may be implementation dependent. Therefore, we list them here explicitly without assigning them a particular value: one is the packet loss detection time (T2), and the other is feedback processing and queuing time for retransmissions (T5). Implementers shall assign appropriate values to these two parameters.
Graphically, we have the following:
Sender
+-+---------------------------------^-----\-----------------
\ \ / \
\ \ | |
SN=0 \ \ SN=1 / \ RTX(SN=0)
\ \ / \
X \ / \
`. / \
\ / \
\ | |
\ / \ ......
\ / \
-------------V----D--------/-----------------------V--------
T1 T2 T3 T4 T5 T1 ........
Receiver
Legend:
- DL: downlink (client->server)
- UL: uplink (server->client)
- Time unit is seconds, s.
- Bitrate unit is bits per second, bps.
DL transmission time: T1 = physical-delay-DL + tx-delay-DL (=avg-pkt-size/DL-bitrate) + interarrival-delay-jitter
Time to detect packet loss: T2 = pkt-loss-detect-time
Time to report packet loss: T3 = time-to-next-rtcp-report
UL transmission time: T4 = physical-delay-UL + transmission-delay-UL + interarrival-delay-jitter
Retransmissions processing time: T5 = feedback-processing-time + rtx-queuing-time
A.2. Goal
To find an estimate of the buffering time, T(), that a streaming server shall use in order to enable a given number of retransmissions for each packet, N. This time is approximately equal at the server and at the client, if one considers that the client starts buffering T1 seconds later.
A.3. Solution
First, we find the value of the estimate for 1 retransmission, T(1)=T:
T = T1 + T2 + T3 + T4 + T5
Since T1 + T4 ~= RTT,
T = RTT + T2 + T3 + T5
The worst case for T3 would be that we assume that reporting has to wait a whole RTCP interval and that the maximum randomization factor of 1.5 is applied. Therefore, after applying the subsequent compensation to avoid traffic bursts (see Appendix A.7 of RTP [3]), we have that T3 = 1.5/1.21828*RTCP-Interval. Thus,
T = RTT + 1.2312*RTCP-Interval + T2 + T5
On the other hand, RTCP-Interval = avg-rtcp-size*8*(senders + receivers)/(RR+RS). In this scenario: sender + receivers = 3; RR+RS is the receiver report plus sender report bandwidth share, in this case, equal to the default 5% of session bandwidth, bw. We assume an average RTCP packet size, avg-rtcp-size = 120 bytes. Thus:
T = RTT + 1.2312*avg-rtcp-size*8*3/(0.05*bw) + T2 + T5
for 1 retransmission.
For enabling N retransmissions, the available buffering time in a streaming server or client is approximately:
T(N) = N*(RTT+1.2312*avg-rtcp-size*8*3/(0.05*bw) + T2 + T5)
where, as per above,
avg-rtcp-size = 120 + (receiver-RTCP-bw-share)*(12 + 4*N)
= 120 + (1/3)*(12 + 4*N)
= 124 + 4*N/3.
A.4. Numbers
If we ignore the effect of T2 and T5, i.e., assume that all losses are detected immediately and that there is no additional delay due to feedback processing or retransmission queuing, we have the following buffering times for different values of N:
RTCP w/ several Generic NACKs; variable packet size = 124 + 4*N/3 bytes
|============|=====|======================================|
| RTP BW | RTT | N value |
|============|=====| 1 2 5 7 10 |
|======================================|
64000 0,05 1,21 2,44 6,28 8,97 13,18
128000 0,05 0,63 1,27 3,27 4,66 6,84
256000 0,05 0,34 0,68 1,76 2,50 3,67
512000 0,05 0,19 0,39 1,00 1,43 2,09
1024000 0,05 0,12 0,25 0,63 0,89 1,29
5000000 0,05 0,06 0,13 0,33 0,46 0,66
10000000 0,05 0,06 0,11 0,29 0,41 0,58
64000 0,2 1,36 2,74 7,03 10,02 14,68
128000 0,2 0,78 1,57 4,02 5,71 8,34
256000 0,2 0,49 0,98 2,51 3,55 5,17
512000 0,2 0,34 0,69 1,75 2,48 3,59
1024000 0,2 0,27 0,55 1,38 1,94 2,79
5000000 0,2 0,21 0,43 1,08 1,51 2,16
10000000 0,2 0,21 0,41 1,04 1,46 2,08
64000 1 2,16 4,34 11,03 15,62 22,68
128000 1 1,58 3,17 8,02 11,31 16,34
256000 1 1,29 2,58 6,51 9,15 13,17
512000 1 1,14 2,29 5,75 8,08 11,59
1024000 1 1,07 2,15 5,38 7,54 10,79
5000000 1 1,01 2,03 5,08 7,11 10,16
10000000 1 1,01 2,01 5,04 7,06 10,08
To quantify the error of not taking the Generic NACKS into account, we can do the same numbers, but ignoring the Generic NACK contribution, avg-rtcp-size ~= 120 bytes. As we see from below, this may result in a buffer estimation error of 1-1.5 seconds (5-10%) for lower bandwidth values and higher number of retransmissions. This effect is low in this case. Nevertheless, it should be carefully evaluated for the particular scenario; that is why the formula includes it.
RTCP w/o Generic NACK, fixed packet size ~= 120 bytes
|============|=====|======================================|
| RTP BW | RTT | N value |
|============|=====| 1 2 5 7 10 |
|======================================|
64000 0,05 1,16 2,32 5,79 8,11 11,58
128000 0,05 0,60 1,21 3,02 4,23 6,04
256000 0,05 0,33 0,65 1,64 2,29 3,27
512000 0,05 0,19 0,38 0,94 1,32 1,89
1024000 0,05 0,12 0,24 0,60 0,83 1,19
5000000 0,05 0,06 0,13 0,32 0,45 0,64
10000000 0,05 0,06 0,11 0,29 0,40 0,57
64000 0,2 1,31 2,62 6,54 9,16 13,08
128000 0,2 0,75 1,51 3,77 5,28 7,54
256000 0,2 0,48 0,95 2,39 3,34 4,77
512000 0,2 0,34 0,68 1,69 2,37 3,39
1024000 0,2 0,27 0,54 1,35 1,88 2,69
5000000 0,2 0,21 0,43 1,07 1,50 2,14
10000000 0,2 0,21 0,41 1,04 1,45 2,07
64000 1 2,11 4,22 10,54 14,76 21,08
128000 1 1,55 3,11 7,77 10,88 15,54
256000 1 1,28 2,55 6,39 8,94 12,77
512000 1 1,14 2,28 5,69 7,97 11,39
1024000 1 1,07 2,14 5,35 7,48 10,69
5000000 1 1,01 2,03 5,07 7,10 10,14
10000000 1 1,01 2,01 5,04 7,05 10,07