[rtcweb] Review Comments on draft-ietf-rtcweb-transports-04.txt
Magnus Westerlund <magnus.westerlund@ericsson.com> Fri, 25 April 2014 14:20 UTC
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Date: Fri, 25 Apr 2014 16:19:52 +0200
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [rtcweb] Review Comments on draft-ietf-rtcweb-transports-04.txt
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Harald and WG, I am catching up on the changes and improvements in the two latest versions. I have tried to do a complete review through and match with my previous comments. First of all, I think the two revisions have improved a number of things. 1. Section 3.1: Platforms that do not give access to these interfaces will not be able to support a conforming RTCWEB implementation. I am very split over this statement. We do want platforms to support per packet DSCP markings, while at the same time we know we might not be able to set DSCP at all for a transport flow (5-tuple). Thus, I wonder if calling the implementations non-conformant due to the underlying platform is a bit though. 2. Section 3.1: This specification does not assume that the implementation will have access to ICMP or raw IP. But, to my understanding we have things that could benefit from getting the information from ICMP, e.g. the SCTP MTU discovery mechanism. Thus, I wonder if this statement should be changed to: This specification does not assume that the implementation will have access to ICMP or raw IP, however access to ICMP information can benefit the WebRTC implementation and its performance. 3. Section 3.4: TURN TCP candidates [RFC6062] MAY be supported. However, such candidates are not seen as providing any significant benefit. First, use of TURN TCP would only be relevant in cases which both peers are required to use TCP to establish a PeerConnection. Secondly, that use case is anyway supported by both sides establishing UDP relay candidates using TURN over TCP to connect to the relay server. Thirdly, using TCP only between the endpoint and its relay may result in less issues with TCP in regards to real-time constraints, e.g. due to head of line blocking. I think these two paragraphs should be moved to after the following paragraph: ICE-TCP candidates [RFC6544] MUST be supported; this may allow applications to communicate to peers with public IP addresses across UDP-blocking firewalls without using a TURN server. >From a discussion of WebRTC RTP or DTLS packets over TCP, it makes better sense to first discuss the ICE TCP candidates, then discuss why TCP candidates from TURN makes less so. 4. Section 3.4: If TCP connections are used, RTP framing according to [RFC4571] MUST be used, both for the RTP packets and for the DTLS packets used to carry data channels. I think this paragraph forgets the encapsulation of the STUN messages that ICE emits for initial path verification and nomination, as well as for keep-alive and consent freshness. 5. section 3.4: NOTE IN DRAFT: This may be added. I don't understand this note. 6. Section 3.5: The setup protocol for RTCWEB data channels is described in [I-D.jesup-rtcweb-data-protocol]. First of all, this pointer is the individual version rather than WG draft. Secondly, I think the DCEP shall be mandated to be supported by an WebRTC implementation in this specification, and thus become a normative reference also. 7. Section 4: The RTCWEB prioritization model is that the application tells the RTCWEB implementation about the priority of media and data flows through an API. As we have discussed before in regards to DART as well as draft-ietf-tsvwg-rtcweb-qos has needs for clear definitions what is meant with media and data flows here. But, reading this section it is clear that the document itself has need for clear definitions. A data flow appear to be the one of the SCTP streams that forms a single data channel, as this is the level where we assign data priority. A WebRTC media flows is a more complex beast. It is one or more RTP packet streams (draft-ietf-avtext-grouping-taxonomy), i.e. the RTP packets associated with an SSRC. However, due to FEC and RTP retransmission as well as scalable codecs, there could be multiple SSRCs that carries data associated and possibly necessary to be able to perform media decoding. Thus, I think we are talking about the RTP packet streams that are the result of a single MediaStreamTrack within one RTCPeerConnection. I would also note that Section 4.1 uses Media Stream instead of media flow. There are need for alignment here. 8. Section 4.1: I think this section is a good start. It clearly needs some more input. Especially about the notes made. In addition to the first three configurations, I think the one where one has different transport flows per priority level for the media flows appears to be one most likely to require mandating implementation support. Yes this means that I also would like to mandate implementation support for the one transport flow per media type configuration. 9. Section 4.1: A receiving implementation MUST be able to receive media and data in all these configurations. It is very unclear what "all" in the above sentence includes. 10. Section 4.2: First of all I do wonder if this is the right solution. I haven't yet spent enough time on the issue to know if the proposal is useful and good enough, or if we need something else. I do encourage people to consider this. However, considering Simon Perrault's comment, I do want to state that I share Harald's view that this prioritization behavior needs to be consistent over the implementations. 11. Section 4.2: This prioritization is independent of the media type, but will lead to packet loss due to full send buffers occuring first on the highest volume flows at any given priority level. I think I have two questions here. First, I appears that you assume that you have an API model where the framework will try to send and fail rather than creating the data based on allowance or with the knowledge that you can send a particular data amount. I think creating sender side buffer overflow is a poor regulation model. Secondly, I would not that this issue has a higher probability of effecting the flow with most data first. Cheers Magnus Westerlund ---------------------------------------------------------------------- Services, Media and Network features, Ericsson Research EAB/TXM ---------------------------------------------------------------------- Ericsson AB | Phone +46 10 7148287 Färögatan 6 | Mobile +46 73 0949079 SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com ----------------------------------------------------------------------
- Re: [rtcweb] Prioritization Harald Alvestrand
- [rtcweb] I-D Action: draft-ietf-rtcweb-transports… internet-drafts
- Re: [rtcweb] I-D Action: draft-ietf-rtcweb-transp… Harald Alvestrand
- [rtcweb] Prioritization Simon Perreault
- Re: [rtcweb] Prioritization Harald Alvestrand
- Re: [rtcweb] Prioritization Simon Perreault
- [rtcweb] Review Comments on draft-ietf-rtcweb-tra… Magnus Westerlund
- Re: [rtcweb] Prioritization Simon Perreault
- Re: [rtcweb] Prioritization Martin Thomson
- Re: [rtcweb] Prioritization Matthew Kaufman (SKYPE)
- Re: [rtcweb] Prioritization Matthew Kaufman (SKYPE)
- Re: [rtcweb] Prioritization Harald Alvestrand
- Re: [rtcweb] Prioritization Cullen Jennings (fluffy)
- Re: [rtcweb] Prioritization Martin Thomson
- Re: [rtcweb] Prioritization Matthew Kaufman (SKYPE)
- Re: [rtcweb] Prioritization Dave Taht
- Re: [rtcweb] Prioritization DRUTA, DAN
- Re: [rtcweb] Prioritization Harald Alvestrand
- Re: [rtcweb] Prioritization Michael Welzl