[rtcweb] [avtext] Payload Types assignments was Re: SV: [mmusic] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11

"Karl Stahl" <karl.stahl@intertex.se> Tue, 22 October 2013 14:38 UTC

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From: Karl Stahl <karl.stahl@intertex.se>
To: 'Harald Alvestrand' <harald@alvestrand.no>, rtcweb@ietf.org, 'Magnus Westerlund' <magnus.westerlund@ericsson.com>
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Subject: [rtcweb] [avtext] Payload Types assignments was Re: SV: [mmusic] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Harald, I mostly agree with the quality requirements of different real-time
traffic that the WebRTC browser/application may use. But rather than asking
the application, let's convey the bandwidth and priority requirements to the
network. Just like with the Payload type (that is hard to squeeze that
information into) it must be visible to the network (and not changed by the
network, like diffserv bits are). Such marking must also be available for
incoming traffic, which is especially important in RSVP type of networks,
that has to reserve bandwidth for it.  


There is actually a good way to show these needs to the network (without
using the PT, or diffserv bits, which aren’t sufficient anyway). 


Let's use the RTP header extension field that also is visible outside the
encrypted payload. A week ago came
http://tools.ietf.org/html/draft-carlberg-avtext-classifier-00  that
outlines the usage of the extension field for classification of traffic!
This document does not yet outline what to put in there and how to encode it


Today's http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10 discusses
other webrtc usages of the RTP header extension in 5.2 (there can be many
header extensions according to RFC 5285) and in 9 there is "WebRTC Use of
RTP: Future Extensions".


So, it looks obvious to use the RTP header extension to show the
characteristics and bandwidth requirements to the network. It should not
introduce any backward incompatibilities either.


Such marking is done in every RTP packet so it can be set individually for
each stream and could even be changed during a session (e.g. when limiting
the bandwidth based on RTCP feedback). RFC 5286 also specifies how RTP
extension header usage can be negotiated in SDP. I think this could be
easily done by the WebRTC browser for "all current and future needs" if
properly specified now.


I suggest that two parameters (e.g. two bytes each) are encoded into the RTP
header extension:


A) The maximum bandwidth requirement: Two bytes could contain everything
from some bps for real-time text to Gbps for future 3D supersize
telepresence… on a logarithmic scale.


B) The quality characteristics for the stream, with the highest bit set to
1, we could allocate a bit each quality e.g:

Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay
Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery,
Prioritize X, Variation Y, that could be combined as required to describe
the stream.


And with highest bit set to 0, there could instead be a number for special
usage that does not fit the general description of the individual bits.


Please note the totally different requirements a diffserv and an RSVP
network have to know, so let’s put all into these bytes. (E.g. a diffserv
network don't need the bandwidth usage, but RSVP reservation networks (e.g.
cable and 3G/4G OTT) do. There one should initially reserve the maximum
bandwidth indicated, but can later re-reserve.)




PS Microsoft seems to have done work in this field, defining a proprietary
attribute “MS Service Quality”; 

However that seems to apply to the TURN server allocation request and would

--- Apply to the whole UDP flow, and could not be set for each stream
individually (with different requirements), and

--- Does not handle the bandwidth requirement for incoming real-time traffic
(required to reserve in RSVP type of networks)

However the quality attributes conveyed and their encoding may  be


This is MS-Service Quality Attribute from 



MS-Service Quality Attribute

The MS-Service Quality attribute is used to convey information about the
data stream that the protocol client is intending to transfer over an
allocated port. The protocol client SHOULD<21> include this attribute as
part of an Allocate request message. A TURN server SHOULD use the
information in this attribute to make decisions about resource allocation,
bandwidth prioritization, and data delivery methods. If the attribute is not
present in the Allocate request message, the TURN server SHOULD assume that
the data stream is audio with best effort delivery. The format of this
attribute is as follows... 


The following stream types are supported in this extension. All other stream
types are reserved for future use.

§ "0x0001": Audio

§ "0x0002": Video

§ "0x0003": Supplemental Video

§ "0x0004": Data

Service Quality (2 bytes): The service quality level required by the
protocol client for the stream.

The following service quality levels are supported in this extension. All
other service quality levels are reserved for future use.

§ "0x0000": Best effort delivery.

§ "0x0001": Reliable delivery.



-----Ursprungligt meddelande-----

Från: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] För Harald

Skickat: den 8 oktober 2013 13:01

Till: rtcweb@ietf.org

Ämne: Re: [rtcweb] Payload Types assignments was Re: SV: [mmusic] WGLC of


On 10/08/2013 09:17 AM, Karl Stahl wrote:

> Hej Magnus,


>> Also, are you really interested in knowing that it is VP9 vs H.264, 

>> isn't

> the questions this is video of this priority that is important?

>> I think you need to more carefully consider what are the goals you 

>> try to

> achieve them.


> Actually, my concern is to get an idea of the maximum bandwidth that 

> could be required for a WebRTC (ICE) setup media flow. Both voice and 

> video should be prioritized over data (their individual priority is of 

> less importance as long as there is sufficient bandwidth for both).


You don't know that without knowing what the application is for.

In, for instance, a shooter game with voice backchannels, the movement and
event information (data) is MORE time sensitive than the voice data.



> With diffserv you don’t need to know the bandwidth requirement, but 

> with RSVP reservation (like in cable and mobile networks) you need to 

> know how much to reserve. Voice is like 100's kbit/s, video VP8 or 

> H.264 is like 3,5 mbps.


Again, without knowing the application, you don't know that.

The application could decide to use QCIF or HD, and the bandwidth variation
of screencast (semi-static with sudden, large changes) is completely
different from that of a talking head, which is again completely different
from a high-movement scene.



> To add to the complication of codec variants, the video codecs in 

> question for WebRTC have variable bandwidth, and when there is a poor 

> connection we see Chrome reducing the video window size to reduce the
bandwidth used...


> I think the payload type field at best can reflect a maximum bandwidth 

> to initially reserve bandwidth for, and thereafter make new 

> reservations if the bandwidth changes during the call. So could we 

> change RTP to show maximum bandwidth instead of payload type in that 

> field outside the encrypted payload :) ... Or maybe that is not a joke?


I think these ruminations only lead to one conclusion:


You can't tell what the needed bandwidth is up front without asking the

You can't tell what the right priority ranking is without asking the


If you need to know the bandwidth or the priority up front, the application
has to tell you. Anything else is pure heuristics.



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