[tram] QoS for RTC over the Internet, DISCUSS: Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs

["Karl Stahl" <[e-mail]>]

Tiru > I did not did not understand how TURN server will identify if it’s WebRTC media streams or gaming traffic or some other data traffic relayed through it to set the diffserv bits correctly !

--- I think you do understand… – but I will spell out that DISCUSS/MALICE does it better and with useful detail J

 

Pål, Tiru, Dan and you other thinking about these things:

 

What has been discussed by me here so far, to give us quality of real-time traffic over the Internet (not a small task) is:

 

1) To direct the real-time traffic to where the network can handle such traffic (using a network offered TURN servers) (which is not within the scope of DISCUSS)

 

2) When such a TURN server flow is allocated, it can ASSUME that it is going to used for real-time traffic and instruct the network (e g via setting diffserve bits) to prioritize the assumed real-time traffic. (Giving the same prioritization to all TURN traffic works quite well. – Only if we fill the whole pipe with prioritized traffic (best effort totally pushed off) is it important that e.g. voice if prioritized higher than video).

 

3) When the TURN server sees the flow coming in (Note: in both directions) it can do smarter guesswork of what traffic it is (like a DPI-box), e.g. what is RTP and what is data channel to instruct the network better.

 

Now after browsing http://tools.ietf.org/html/draft-martinsen-tram-discuss-00 

 

4) DISCUSS transfers information directly from the application to the network at the time of setup of the STUN or TURN(?) server, which it therefore can do with better detail and prediction. This is valuable, also for reserving bandwidth in non diffserve networks like Cable and  Mobile where one reserves bandwidth rather than use diffserve for QoS.

 

5) Isn’t DISCUSS usable with TURN? Maybe even better! And it can be used with 1) above J

You can transfer the same information in the TURN allocate request as in the STUN binding request, can’t you?. I searched for “TURN” in the DISCUSS draft, but could not see it spelled out. TURN is an extension to STUN, so maybe it is just obvious? – I have not checked details in the specs so Pål, Tiru or Dan knowing better, please confirm or correct!

 

Some observations:

 

a. In DISCUSS using STUN, the network element doing the diffserv or reservation settings WOULD be in the default gateway.

b. In DISCUSS using TURN, the TURN server doing the diffserv or reservation settings COULD be in the default gateway. (The auto-discovery of the TURN server would simply point out the default gateway.)

 

Sound very similar: Could not DISCUSS over TURN always be used?

 

c. With DISCUSS using TURN, the application would directly talk to the network device doing the diffserve settings etc. (instead of through it, where typically the default gateway would snope that talk). Would that not easy some of the concerns in the draft left for further discussion? 

 

d. Can we add info in the response? E.g. if the network device is only willing to give 1 Mbps instead of needed 3 Mbps, so the application can be informed to reduce his video resolution? Would be a nice mechanism when RTC alone starts filling our pipes. (I saw a similar idea by Pål in the previous email I just commented.)

 

 

6) Please add to the DISCUSS draft that it also could reserve bandwidth in bandwidth reservation type of networks like Cable and  Mobile networks!

 

 

A few more things are needed for the ultimate goal, bringing end-to-end QoS or QoE for real-time communication to Best Effort Internet (which does not seem impossible, but quite doable now J ) remains though. I’ll come back to those. 

 

It will e.g. relate to how to do with INCOMING traffic, especially in reservation type of networks, and the wild changing/stripping of diffserve bits between ISPs.

 

You may want to check this old discussion to see if this useful:

http://www.ietf.org/mail-archive/web/rtcweb/current/msg09128.html 

http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html 

 

/Karl

 

 

Från: Tirumaleswar Reddy (tireddy) [mailto:[e-mail]] 
Skickat: den 13 februari 2014 17:47
Till: Karl Stahl
Kopia: [e-mail]
Ämne: RE: IMPORTANT CLARIFICATIONS: [tram] Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs

 

Hi Karl,

 

I did not understand how TURN server will identify if it’s WebRTC media streams or gaming traffic or some other data traffic relayed through it to set the diffserv bits correctly !

 

-Tiru.

From: Karl Stahl [mailto:[e-mail]] 
Sent: Thursday, February 13, 2014 5:05 PM
To: Tirumaleswar Reddy (tireddy); 'Hutton, Andrew'; 'Justin Uberti'; Muthu Arul Mozhi Perumal (mperumal)
Cc: [e-mail]; 'Simon Perreault'; 'Oleg Moskalenko'; [e-mail]; 'Marc Blanchet'; Dan Wing (dwing)
Subject: IMPORTANT CLARIFICATIONS: [tram] Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs

 

On the side of this TRAM-list, I also got this question:

> Regarding the enterprise case, I am not sure I follow your argument.

> Do you mean that by setting up an enterprise TURN server, and open the firewall for media over UDP from/to TURN server be the solution?

---- As we all realize, that would of course not help or improve things

 

The intended solution in the enterprise case has not yet been spelled out in this TRAM-list discussion, so for better understanding, let me copy a few things from the discussion in September/October on the RTCWEB-list and what is (since long) spelled out in the draft-ietf-rtcweb-use-cases-and-requirements.

 

For better understanding, I also want to point out that a TURN can have two interfaces (acting like a router for media between different networks). This allows to easier understand that can TURN servers can direct a best media path (rather than just thinking that a TURN service is a device which media just bounces against at one interface). 

 

And, we can also hope for that a TURN server becomes a (common) component of a firewall, which would allow the firewall to understand that the media directed to it is RTC and should be prioritized whereby the firewall can traffic shaped (back-off data traffic that may be filling its Internet pipe) as well as e.g. set diffserve bits or take other measures to assist proper quality handling thought the network. (These are common mechanisms available and used in firewalls/NATs/access routers, but TURN servers are not yet included such devices.) The same goes for access routers/default gateways, DPIs in the transport network itself – TURN servers included in such points were media can pass and quality measures applied may/will be very useful to get us WebRTC media with through networks without quality destruction.

 

>From the RTCWEB mailing list September 20th (by me):

An enterprise network that want to keep a restrictive firewall not allowing UDP traffic, could provide a real-time path using a TURN server paralleling the firewall, instead of tunneling RTP through always open http or https ports resulting in RTP media over TCP – with severe quality problems from TCP retransmissions of dropped packets. The TURN server address is most easily provided in the same way as the IP address and DNS address. (That would also put the right party in control – The network provider (here the enterprise) decides what is allowed on his network.)

 

The browser should select which available TURN server address to use in the following priority order, where ICE could be used to try several:

 

1) TURN server address configured in the browser by the user (special cases, normally not used)

2) TURN server address configured by the network administrator via an “admin policy template”

3) TURN server address supplied by DHCP or similar automatic network method

4) TURN server address being supplied by the web application"

 

 

And from yesterdays(!) http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-14 these enterprise things and necessity are spelled out in:

 

F19     The browser must be able to use several STUN and TURN servers

   ----------------------------------------------------------------

 

A22

 <http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-14#section-3.3.5> 3.3.5.  Simple Video Communication Service, enterprise aspects

 <http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-14#section-3.3.5.1> 3.3.5.1.  Description

   This use-case is similar to the Simple Video Communication Service

   use-case (Section 3.3.1 <http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-14#section-3.3.1> ).

 

   What is added is aspects when using the service in enterprises.  ICE

   is assumed in the further description of this use-case.

 

   An enterprise that uses a RTCWEB based web application for

   communication desires to audit all RTCWEB based application sessions

   used from inside the company towards any external peer.  To be able

   to do this they deploy a TURN server that straddles the boundary

   between the internal and the external network.

 

   The firewall will block all attempts to use STUN with an external

   destination unless they go to the enterprise auditing TURN server.

   In cases where employees are using RTCWEB applications provided by an

   external service provider they still want the traffic to stay inside

   their internal network and in addition not load the straddling TURN

   server, thus they deploy a STUN server allowing the RTCWEB client to

   determine its server reflexive address on the internal side.  Thus

   enabling cases where peers are both on the internal side to connect

   without the traffic leaving the internal network.  It must be

   possible to configure the browsers used in the enterprise with

   network specific STUN and TURN servers.  This should be possible to

  achieve by auto-configuration methods.  The RTCWEB functionality will

   need to utilize both network specific STUN and TURN resources and

   STUN and TURN servers provisioned by the web application.

 <http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-14#section-3.3.5.2> 3.3.5.2.  Additional Requirements

   ----------------------------------------------------------------

   REQ-ID      DESCRIPTION

   ----------------------------------------------------------------

   F20     The browser must support the use of STUN and TURN

           servers that are supplied by entities other than

           the web application (i.e. the network provider).

   ----------------------------------------------------------------

 

There are further requirement listed, helping us to understand the need for auto discovery and a network provided TURN-server should be used to ENFORCE that media takes that path (and thus, other paths e.g. suggested by the remote MUST not happen to be used). This is related to the mobility aspect (valid even without the roaming idea):


 <http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-14#section-3.3.6> 3.3.6.  Simple Video Communication Service, access change


 <http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-14#section-3.3.6.1> 3.3.6.1.  Description

   This use-case is almost identical to the
   Simple Video Communication Service use-case (Section 3.3.1 <http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-14#section-3.3.1> ).  The
   difference is that the user changes network access during the
   session.
 
   The communication device used by one of the users has several network
   adapters (Ethernet, WiFi, Cellular).  The communication device is
   accessing the Internet using Ethernet, but the user has to start a
   trip during the session.  The communication device automatically
   changes to use WiFi when the Ethernet cable is removed and then moves
   to cellular access to the Internet when moving out of WiFi coverage.
   The session continues even though the access method changes.

 <http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-14#section-3.3.6.2> 3.3.6.2.  Additional Requirements

   ----------------------------------------------------------------
   REQ-ID      DESCRIPTION
   ----------------------------------------------------------------
   F17     The communication session must survive across a
           change of the network interface used by the
           session
   ----------------------------------------------------------------

 <http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-14#section-3.3.7> 3.3.7.  Simple Video Communication Service, QoS


 <http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-14#section-3.3.7.1> 3.3.7.1.  Description

   This use-case is almost identical to the
   Simple Video Communication Service, access change use-case
   (Section 3.3.6 <http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-14#section-3.3.6> ).  The use of Quality of Service (QoS) capabilities is
   added:
 
   The user in the previous use case that starts a trip is behind a
   common residential router that supports prioritization of traffic.
   In addition, the user's provider of cellular access has QoS support
   enabled.  The user is able to take advantage of the QoS support both
   when accessing via the residential router and when using cellular.

 <http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-14#section-3.3.7.2> 3.3.7.2.  Additional Requirements

   ----------------------------------------------------------------
   REQ-ID      DESCRIPTION
   ----------------------------------------------------------------
   F17     The communication session must survive across a
           change of the network interface used by the
           session
   ----------------------------------------------------------------
   F22     The browser must be able to receive streams and
           data from multiple peers concurrently.
   ----------------------------------------------------------------

 

Further from the RTCWEB mailing list September 20th (by me):

 

There are several reasons for a network service provider to supply a TURN server as part of his offered access:

- to keep media paths short, specifically not sending media outside its own network to some distant application provided TURN server

- to support mobility, i.e. you may want to move from a LAN with a configured TURN server to accessing via WiFi or 3G/4G OTT channels

- to offer a media path with better quality (than best effort data traffic).

Getting “WebRTC-ready” access and we look forward to telepresence for everyone.

 

 

I hope this (a bit lengthy) summary of already thought-out and discussed aspects/requirements will help us understand that the auto-discovered TURN server is an ORDER from the enterprise and/or the NSP/ISP  to send the media through this TURN path, and that other media paths that may exist MUST NOT BE USED. (That is why we especially have to watch/advice that workable media paths proposed by the remote party not becomes used “by accident”.

 

/Karl

 

 

Från: Tirumaleswar Reddy (tireddy) [mailto:[e-mail]] 
Skickat: den 13 februari 2014 04:37
Till: Hutton, Andrew; Justin Uberti; Muthu Arul Mozhi Perumal (mperumal)
Kopia: [e-mail]; Simon Perreault; Oleg Moskalenko; [e-mail]; Marc Blanchet; Dan Wing (dwing); Karl Stahl
Ämne: RE: [tram] Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs

 

Hi Andy,

 

There are other ways to solve the problem for example using PCP. Can you clarify how deploying a TURN server in the Enterprise protects the users and the network ? 

 

-Tiru.

From: tram [mailto:[e-mail]] On Behalf Of Hutton, Andrew
Sent: Thursday, February 13, 2014 1:00 AM
To: Justin Uberti; Muthu Arul Mozhi Perumal (mperumal)
Cc: [e-mail]; Simon Perreault; Oleg Moskalenko; [e-mail]; Marc Blanchet; Dan Wing (dwing); Karl Stahl
Subject: Re: [tram] Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs

 

The case where the TURN server is the only option may become common within enterprise networks and that might be deliberate enterprise policy because it provides the better path (UDP through the F/W) and protects the users and the network.

 

Andy

 

 

From: tram [ <mailto:[e-mail]> mailto:[e-mail]] On Behalf Of Justin Uberti
Sent: 12 February 2014 17:46
To: Muthu Arul Mozhi Perumal (mperumal)
Cc:  <mailto:[e-mail]> [e-mail]; Simon Perreault; Oleg Moskalenko;  <mailto:[e-mail]> [e-mail]; Marc Blanchet; Dan Wing (dwing); Karl Stahl
Subject: Re: [tram] Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs

 

Agree. If TURN is indeed being provided for the user's benefit, the client's ICE logic (based on RTT or similar) should result in it preferring the TURN path.

 

On Wed, Feb 12, 2014 at 1:27 AM, Muthu Arul Mozhi Perumal (mperumal) <[e-mail]> wrote:

Yes, I believe the second case is rare, but would be better than a rat race b/w administrators trying to block p2p traffic and force it through a TURN server and apps/endpoints finding smarter ways to bypass them.

 

Muthu

 

From: Oleg Moskalenko [mailto: <mailto:[e-mail]> [e-mail]] 
Sent: Wednesday, February 12, 2014 1:07 PM
To: Muthu Arul Mozhi Perumal (mperumal)
Cc: Justin Uberti; Karl Stahl;  <mailto:[e-mail]> [e-mail]; Marc Blanchet;  <mailto:[e-mail]> [e-mail]; Dan Wing (dwing); Simon Perreault


Subject: Re: [tram] Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs

 

The TURN server has to be used when it is either the only option, or if it provides a better path (I guess the second case is rather rare).

 

On Tue, Feb 11, 2014 at 11:32 PM, Muthu Arul Mozhi Perumal (mperumal) <[e-mail]> wrote:

+1

 

Forcing all traffic through a TURN server and expecting it would provide the best user experience doesn't look the right approach. Instead, if a path through a TURN server exists and does provide lower RTT, jitter etc, being able to detect and use (or switch to) that path might be desirable..

 

Muthu

 

From: tram [mailto: <mailto:[e-mail]> [e-mail]] On Behalf Of Justin Uberti
Sent: Wednesday, February 12, 2014 11:43 AM
To: Karl Stahl
Cc:  <mailto:[e-mail]> [e-mail]; Marc Blanchet;  <mailto:[e-mail]> [e-mail]; Dan Wing (dwing); Simon Perreault
Subject: Re: [tram] Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs

 

Inline.

 

On Tue, Feb 11, 2014 at 2:37 PM, Karl Stahl <[e-mail]> wrote:

Listening to this thread, I am afraid we are missing the very point and necessity for this milestone!

- There are severe NAT traversal and quality issues that should and can be dealt with by a good auto-discovery mechanism and the right usage by the turn client (the WebRTC browser)

 

There are ways, not only: Enterprises or ISPs wishing to provide their own TURN server, in an attempt to reduce so-called "triangle routing",need a new auto-discovery mechanism

But also: - NSPs (Network Service Providers) want to provide a path where the bandwidth of WebRTC is better coped with.

- NSPs or Enterprises want to offer an Internet access quality pipe for prioritized RTC (Real Time Communication) traffic. 

- Enterprises having restrictive firewalls, want to provide a UDP-path for WebRTC and possibly also for better quality where RTC do not compete with data traffic. 

Also considering

- Mobility; It is common to move from a LAN to accessing via WiFi or 3G/4G OTT channels, all should be able to automatically offer their own optimal TURN server

 

This leads us into  “TURN…to identify WebRTC flows” etc! It is not a mistake, but the very need for this milestone!

 

Again, it has not been demonstrated why TURN is the right technology here, compared to a more transparent flow identification tool like MALICE. We don't force all HTTP requests to locate a HTTP proxy via anycast, I don't see why we need to do the same for WebRTC. 

 

What are the hesitations raised here?

> TURN primarily to identify WebRTC flows, as opposed to using it as a NAT traversal tool. This makes me concerned that we may be using the wrong technology to solve the problem

It is correct that ICE/STUN/TURN was designed to address the NAT/Firewall traversal problem associated with real-time communication (SIP at that time). However, its largest flaw/problem is that quality things were not (could not be?) considered. The method’s very idea (like all similar methods for getting RTC through ordinary NAT/Firewalls) is to fool the media through a NAT/Firewall that is unaware of what is happening. Thus, this is root of quality issues (and bandwidth allocation optimization) that needs to be dealt with: Real-time traffic fighting with a data traffic crowded congestion point.

 

I think that "fooling" is an incorrect description. The NAT is supposed to be transparent to the client.

 

But, a BLESSING of ICE/STUN/TURN is that it can be seen as a legitimate request for a suitable pipe for quality demanding real time traffic. J 

ICE is a pre-protocol you use because you want a path for real-time media between parties. Here: The browser says knock knock, I want to get media through (and of course with as good quality as required and possible). 

 

If the NAT/Firewall owner and network owner are allowed to see these requests, they can help/assist in achieving the good media path. If they are not aware, they cannot help!

 

Hope this made it understandable on an overview level how this can become “TURN…to identify WebRTC flows”

It is also the ONLY way I can see to achieve what we want to achieve and should be the aim and requirement of this milestone.

 

I am talking about general usage of WebRTC over Internet/mobile OTT (not feeding WebRTC into application specific networks like IMS where other methods may exist). 

 

This is good, not evil!

 

If the hesitations are raised because of a belief/hope/wish that there are no or will not be severe quality issues “because it is all about bandwidth”, “it will resolve itself with time” etc., I strongly object! That is wrong and will be very detrimental for WebRTC usage. We already see it and I can give numerous examples of how much less quality demanding VoIP is/is not handled quality wise and that it matters. And, what would be bad considering quality issues and allowing/encouraging methods to deal with them?

 

If the hesitations are raised, because of suspicion that the methods we may recommend may be misused to stop/block/destroy WebRTC usage (e.g. to protect income from carrier telephony traffic), I could understand and would fight the same battle. But hopefully, those days are (soon) over – At least forward thinking carrier’s realize that already. Web RTC will happen. Which customers want to pay for an access with blocked WebRTC? The carrier’s offering/assuring good WebRTC will rather get the customers and income J. (Maybe the Web browser can detect and encourage this…) 

 

If there are technical concerns of bad result, or better methods allowing network providers and LAN managers to offer and inform the browser that there are good media paths to be used, and that the web browser automatically can chose those, then let us all understand those, so we can achieve what should be achieved by this milestone.

 

Skype, Hangouts, Facetime are doing billions of minutes per week and the Internet has not melted yet. If we need to do flow identification to allow traffic to be prioritized, fine (see above regarding my preferred approach), but forcing all WebRTC traffic through a MITM (TURN server) is a much bigger jump that I don't yet see the justification for.

 

In short: TURN is a technology that is supposed to fade away with the move to IPv6. I don't think we want to make it a critical element of WebRTC.

 

/Karl

 

 

Från: Dan Wing [mailto: <mailto:[e-mail]> [e-mail]] 
Skickat: den 11 februari 2014 18:25
Till: Marc Blanchet
Kopia: Justin Uberti;  <mailto:[e-mail]> [e-mail]; Karl Stahl;  <mailto:[e-mail]> [e-mail]; Simon Perreault


Ämne: Re: [tram] Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs

 

 

On Feb 11, 2014, at 9:08 AM, Marc Blanchet < <mailto:[e-mail]> [e-mail]> wrote:

 

Le 2014-02-11 à 00:39, Dan Wing < <mailto:[e-mail]> [e-mail]> a écrit :

 


On Feb 10, 2014, at 5:30 PM, Justin Uberti < <mailto:[e-mail]> [e-mail]> wrote:

 

Good to see there is a lot of interest for this milestone. But based on the description here, it seems like we want to use TURN primarily to identify WebRTC flows, as opposed to using it as a NAT traversal tool. This makes me concerned that we may be using the wrong technology to solve the problem.

 

+1.

 

I would prefer allowing flows to establish themselves using their 'best' path, and the best path is seldom through a TURN server.  When we imagine IPv6 in our future, we don't want to force an application-level proxy (TURN) server on the path solely for traversing an IPv6 firewall.

 

 

It seems this thread is conflating all the possible reasons / justifications for TURN:

  * mobility

  * NAT traversal (both endpoints are behind endpoint-dependent mapping NATs)

  * firewall traversal (firewall blocks UDP)

  * enhancing privacy

 

Unfortunately the TURN server nor the endpoint really know which of those use-cases is desired (by the user or by the IT network administrator) or necessary (for the call to work at all). 

 

Dan, while I agree in principle, I doubt that a user could ever say "I want mobility or I want NAT traversal". I think the user only want the call to succeed, whatever the properties of its network point of attachment are.

 

So what can we do?  Should the TURN server provide any and all services the TURN client might possibly want, as that is what a robust TURN server will do, and the endpoint should prefer TURN candidates over all others because there might be some functionality / usefulness of TURN that the user might gain through TURN (e.g., enhanced privacy)?

 

-d

 

 

 

 This seems problematic.  Perhaps we need a way to signal the desired use-case ("trait"), or as Justin suggests, using a different technology for some of these use-cases.

 

-d

 

 

 

 

On Mon, Feb 10, 2014 at 3:18 PM, Karl Stahl < <mailto:[e-mail]> [e-mail]> wrote:

Simon,

Good questions - see inline below --> .
Some more thought is required!

/Karl

-----Ursprungligt meddelande-----
Från: tram [mailto: <mailto:[e-mail]> [e-mail]] För Simon Perreault
Skickat: den 10 februari 2014 15:16
Till: Karl Stahl;  <mailto:[e-mail]> [e-mail];  <mailto:[e-mail]> [e-mail]
Ämne: Re: [tram] Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs


Karl,

It is great to see such enthusiasm! Thanks!

I have a couple technical questions...

Le 2014-02-08 08:11, Karl Stahl a écrit :
> - Note that to achieve some of the above points, TURN must be favored
> over STUN to enforce that the TURN-path actually is used. (The Anycast
> method suggested below, “automatically” does this.)

I understand the STUN vs TURN priority issue. But I don't see how anycast affects it in any way. Can you please explain?

--- Good point - I was a bit quick here (maybe too quick)
We have given this quite bit of thought, since even if a TURN server is provided and discovered, CURRENT usage of ICE may suggest a candidate from the remote party that will make a connection without the need/usage of the TURN server (that we wanted to be used for the good purposes listed).

The only way we found around this, was to stop STUN through the IP default gateway (like a restrictive Enterprise firewall does inhibiting ICE connectivity, which others are concerned about...). Since the provisioning of auto-discovery using the anycast mechanism, would be adding a route in a default gateway, adding a firewall rule to eat STUN packets would assure that the provisioned TURN server actually becomes used (and not bypassed "by accident"). (That was the thought behind the “automatically” within quotes.)

BUT, since you brought up the question, assuming that we have the power to enforce WebRTC usage of ICE, I believe a MUST requirement to use an auto-discovered TURN server instead of STUN, would solve the same problem. However, thinking further (in relation to your next question - "anyone could set up a badly-maintained" - enforcing such ICE usage may not be good.)



> - 3^rd The Anycast method below – I see no problem
>
> It also has the advantage of encouraging (but not requiring) the
> STUN/TURN to be built in the default gateway or NAT/firewall/access
> router itself, with a second interface to a public IP address on the
> WAN side. (Current volume deployed, low cost NSP triple play modems
> usually have a quality assured level 2 or level 3 WAN pipe for just
> voice (and another for IPTV) – The anycast discovered TURN-server can
> be the access gateway to such quality pipe for WebRTC media, in a
> single NSP provided CPE, scaling from residential and up.)

Suppose we define well-known anycast TURN server addresses. How would this not be subject to the same service quality issues that plagued 6to4? That is, anyone could set up a badly-maintained, under-provisioned TURN server and announce it over BGP to the world, as it was done for
6to4 relays. Or just bad BGP outbound filter configuration. And how can we prevent triangle routing? There is nothing guaranteeing that the anycast server you see is being provided to you by your ISP, rather than a server sitting on the other side of the planet.

--- Good point - needs to be resolved. For this I don't have a ready answer...
An auto-discovered TURN server must be trusted (whatever method it is discovered by). We are trusting the one providing us with an IP address and default gateway anyway. It would be easy if we could reuse that trust, instead of another mechanisms.

Is there a good way for the browser to check that the anycast address is not handled beyond the network service provider's default gateway? Ideas?




Thanks,
Simon
--
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NAT64/DNS64 open-source        -->  <http://ecdysis.viagenie.ca/> http://ecdysis.viagenie.ca
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