Subject: Re: [AVTCORE] Possible Additional functionalities for draft-brandenburg-avt-rtcp-for-idms

From: Magnus Westerlund <[email-masked]>
Date: Wed, 2 Nov 2011 10:55:48 +0100

On 2011-10-19 20:56, Fernando Boronat Seguí wrote:
> Magnus,
> 
> Yes, you are right. That is our purpose.
> 
> What do you think about our proposal? Any suggestion will be much
> appreciated.

I do think the basic problem is a relevant one for any use cases where
having less than perfect playout sync at startup is not acceptable. For
example for the distributed loud speaker use case.

Then how common that use case is. And if the simplest way of resolving
this is simply to mandate the end-point to support the RTCP XR DLRR (RFC
3611) mechanism to enable them to determine all the transport delays
between the nodes.

Cheers

Magnus

> 
> Thank you for your response.
> 
> 
> El 19/10/2011 16:41, [email-masked] escribió:
>> Date: Wed, 19 Oct 2011 11:15:40 +0200
>> From: Magnus Westerlund <[email-masked]>
>> To: [email-masked]
>> Subject: Re: [AVTCORE] Possible Additional functionalities for
>> 	draft-brandenburg-avt-rtcp-for-idms
>> Message-ID: <[email-masked]>
>> Content-Type: text/plain; charset="windows-1252"
>>
>> Hi,
>>
>> If I understand the main issue you have is to determine the set of
>> transmission delays from a given media source to its population of
>> receivers so that an appropriate playout time can be selected by the
>> media source prior to actually having sent any media?
>>
>> Cheers
>>
>> Magnus
>>
>> On 2011-10-11 15:37, Fernando Boronat Segu? wrote:
>>> Dear Group members,
>>>
>>> As this is our first mail to the list, we introduce ourselves. We are 
>>> Fernando Boronat and Mario Montagud, co-authors of the Internet Draft 
>>> (I-D) ?RTCP for inter-destination media synchronization? 
>>> (draft-ietf-avtcore-idms-01.txt).
>>> As the I-D was accepted as a WG document, any changes will have to be 
>>> approved by the AVTCORE group. So, we would like to know if you consider 
>>> appropriate to introduce some additional functionalities to the proposed 
>>> IDMS approach in the I-D.
>>>
>>> In many distributed applications it is required that all the receivers 
>>> start the playout of the media stream almost simultaneously (we refer 
>>> this as Coarse Sync) and, after that, all of them play-out the media in 
>>> a synchronous way (we refer this as Fine Sync). Those processes are 
>>> essential for acquiring IDMS. A Global Initial Playout Instant for all 
>>> the geographically distributed receivers (SCs) in a media session could 
>>> prevent from an initial playout time discrepancy (asynchrony) at the 
>>> beginning of the playout of the media stream/s (if a fixed global 
>>> initial buffering delay is set) mainly due to the variable network 
>>> delays from the media server to each one of the SCs.
>>>
>>> Thus, we are thinking about including a new section in the I-D about 
>>> possible packets for Coarse Sync for both the initial playout instant 
>>> (at the begining of a media session) and the initial instant at each 
>>> stage in which the session can be divided (we consider that a stage ends 
>>> when no streams are transmitted during a fixed interval (it could be 
>>> defined depending on the applications), and that the next stage starts 
>>> when data is transmitted again).
>>>
>>> This technique has been considered in some previous IDMS works (see 
>>> reference [Boronat09] in the I-D), but using proprietary protocols. The 
>>> methods we propose here are based on the use of already defined and 
>>> standard RTCP packets, which would avoid the introduction of proprietary 
>>> closed solutions.
>>>
>>> Regarding the Initial Playout Instant, we have primarily thought about 
>>> the use of already proposed RTCP XR packets (RFC 3611):
>>>
>>> 1. Use of RTCP XR with RRTR and DLRR report blocks (defined RFC 3611) 
>>> for measuring the delay between MSAS and SCs, allowing MSAS to estimate 
>>> the RTT for all the SCs, then calculate a common initial playout instant 
>>> and then communicate it to all SCs by sending them an RTCP IDMS Packet 
>>> (defined in the I-D). The RTCP IDMS packet must include the RTP 
>>> Timestamp of the first MU to be sent by the source and its proper 
>>> playout instant, which will be common for all of them.
>>> MSAS may estimate its round trip time (RTT) to the SCs by sending them 
>>> an RTCP XR with the Receiver Reference Time Report Block and forcing 
>>> (according to RFC 3611) them to reply by sending an RTCP XR with a DLRR 
>>> Report Block. Notice that both XR blocks are defined in RFC 3611 to be 
>>> used in the opposite way we propose.
>>>
>>> If this solution is not considered appropriate we could think about 
>>> another solution based on the packets already defined in the I-D, but 
>>> maybe we would need to change the format adding new field/s.
>>>
>>> 2. Once joined the session, if it is not started yet, all the SCs could 
>>> start sending RTCP XR with IDMS block (until they receive the Initial 
>>> Playout Instant indication, sent using the RTCP IDMS Packet) as it is 
>>> defined in the I-D, but including only the essential timing information 
>>> (e.g. the wallclock time at which the packet is sent by the SC) to let 
>>> the MSAS estimate the one-way network delays (not round-trip as in the 
>>> previous case) for each SC (assuming symmetric delays as in TCP) and 
>>> calculate the Initial Playout Instant. For that purpose, some field of 
>>> the RTCP XR IDMS block should be used to indicate the XR IDMS block is 
>>> carrying out temporal information for Coarse Sync, and not for reporting 
>>> about arrival/playout timing of a specific RTP packet.
>>>  From the Initial Instant they will continue playing the streams and 
>>> sending the RTCP XR with IDMS complete block (as currently defined in 
>>> the I-D). It could be a possible solution.
>>>
>>> The process of the calculation of the optimum Initial Playout Instant 
>>> must take into consideration some factors such as the maximum and 
>>> minimum estimated delays, jitter values, expected duration of the 
>>> session, or probable clock deviations, in order to allow enough buffered 
>>> MUs to start the playout and to maintain it continuously, thus avoiding 
>>> underflow/overflow situations (playout discontinuities), during the 
>>> session. This process is out of the scope of the ID.
>>>
>>> With the introduction of the Coarse Sync functionality, the IDMS 
>>> proposal can counteract the effect of the delay variability at the 
>>> beginning of the playout of the media stream/s in each one of the stages 
>>> in which a media session can be divided.
>>>
>>> We prefer the first method because there is no need to define new IDMS 
>>> control messages (or to include some fields to already defined packets), 
>>> but the use of already defined RTCP reports (RFC 3611) can be adapted 
>>> for that purpose.
>>>
>>> Moreover, with the use of the first method, SCs do not have to send 
>>> control messages once they join the session, which may possibly result 
>>> on inefficient control traffic overhead. They will only have to send 
>>> RTCP XR DLRR messages in response to a new incoming RTRR message from 
>>> the source, which is who really knows the initial transmission time for 
>>> the RTP media stream. Moreover, the Initial Playout Instant can be sent 
>>> to the receivers by using the RTCP IDMS packet, already introduced in 
>>> the I-D.
>>>
>>> Feedback will be much appreciated.
>>>
>>> Regards
>>> ========================================================================
>>> Dr. Fernando Boronat Segu?
>>> IEEE Senior Member
>>> Profesor Titular / Lecturer
>>> Dept. Comunicaciones Tlf.-+34+962 849 300 Ext.-49341
>>> Tlf. directo -+34+962 849 341 Fax.-+34+962 849 309 (Compartido)
>>> UNIVERSIDAD POLIT?CNICA DE VALENCIA-CAMPUS DE GANDIA
>>> Calle Pararaninfo, 1, CP. 46730, Grao de Gandia (Valencia), SPAIN
>>> ========================================================================
>>> _______________________________________________
>>> Audio/Video Transport Core Maintenance
>>> [email-masked]
>>> https://www.ietf.org/mailman/listinfo/avt
>>>
>>
> 
> -- 
> Saludos/Regards
> ========================================================================
> Dr. Fernando Boronat Seguí
> Profesor Titular / Lecturer	
> Dept. Comunicaciones           Tlf.-+34+962 849 300 Ext.-49341
> Tlf. directo -+34+962 849 341  Fax.-+34+962 849 309 (Compartido)
> UNIVERSIDAD POLITÉCNICA DE VALENCIA-CAMPUS DE GANDIA
> Calle Pararaninfo, 1, CP. 46730, Grao de Gandia (Valencia), SPAIN
> ========================================================================
> 

-- 

Magnus Westerlund

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---

From: Fernando Boronat Seguí <[email-masked]>
Date: Wed, 02 Nov 2011 11:47:06 +0100

Dear Magnus,

Thank you for your answer.

After some discussion about it, in a recent post to the list we have 
suggested two possible valid methods to be discussed (one proposed by me 
and other proposed by Kevin Gross). Ray Brandenburg will present them in 
Taipei IETF meeting to get some feedback (if he has enough time).

We discarded the use of RTCP XR DLRR reports, as I proposed at first, 
because they will be useful to obtain RTT but no one-way delay we are 
interested in.

Regards
Fernando Boronat

El 02/11/2011 10:55, Magnus Westerlund escribió:
>>> From: Magnus Westerlund<[email-masked]>
>>> To: [email-masked]
>>> Message-ID:<[email-masked]>