Re: [Sip] Silence Suppression in SDP for VoIP

[Qiru Zhou <qzhou@lucent.com>]

In the following cases, there are needs to to understand the far end device
attributes:

1. A device connect to a non human device, such as a speech recognition
   receiver.
   In this case, the non human device may not work since it may less
   tolerant than human. The current automatic detection algorithms may
   not be reliable enough to cover all situations.

2. The channel is used to pass non voice signals (such as fax) and in-band
   control signals.
   In this case, the non human device may not work due to the adaptation
   algorithm signal distortion.

3. A device connect to a gateway with network (far end) echo canceller/noise
   suppresser.
   In this case, the communication still work but may have degraded
   performance due to the incorrect estimation of the echo canceller/noise
   suppresser parameters.

It will be desirable to exchange device attribute information such as
echo canceller/noise suppresser at the session setup time than guess them
remotely. It looks natural to me to use SDP "ecan" and "silenceSupp" defined
in RFC 3108 in VoIP and converged networks.

-- Qiru Zhou

Subject: 
       RE: [Sip] Silence Suppression in SDP for VoIP
   Date: 
       Tue, 4 Feb 2003 13:01:28 -0500
  From: 
       "Steve Silverman" <steves@shentel.net>
    To: 
       "Alex Agranov" <sagranov@COMGATES.co.il>, "'David R. Oran'" <oran@cisco.com>, <sip@ietf.org>


I would have thought that if one side is using silence suppression, the other side
needs to understand this.  If true, how is the use of this communicated?  
 
Steve Silverman

     -----Original Message-----
     From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Alex Agranov
     Sent: Tuesday, February 04, 2003 12:30 PM
     To: 'David R. Oran'; 'sip@ietf.org'
     Subject: RE: [Sip] Silence Suppression in SDP for VoIP

     Hi, 

     I see your point. 
     But if this is right, then why does H323 protocol support silence suppression and echo 
     cancellation parameters? 

     On the other hand, in case of fax fallback to G711 coder (if T38 is not supported), there is a 
     need to specify to remote device that "echo cancelation is ON, and silence suppression is OFF". 
     AFAIK fax will not be properly passed if silence suppression is not turned off. 
     And this, I beleive, is the reason why draft-ietf-sipping-realtimefax-00.txt makes use of 
     "ecan" and "silenceSupp" SDP attributes. 

     Also there is a following sentence in H323-SIP interworking draft : 
             "A fmtp SDP attribute for silence suppression SHOULD be defined if 
             silence suppression is on." 
     Is it not valid any more? 

     Best regards, 
                  Alex Agranov 

     > -----Original Message----- 
     > From: David R. Oran [mailto:oran@cisco.com] 
     > Sent: Tuesday, February 04, 2003 5:53 PM 
     > To: Alex Agranov; 'sip@ietf.org' 
     > Subject: Re: [Sip] Silence Suppression in SDP for VoIP 
     > 
     > 
     > There are no such parameters in SDP nor SIP because both of these are 
     > algorithm options for one end of a media stream. The source 
     > of the stream 
     > does these by itself and therefore they have no end-to-end 
     > significance. As 
     > a consequence there is no need for the receiver to know 
     > anything about Vad 
     > or ecan parameters of the transmitter, or vice versa. 
     > 
     > The reason these exist in MGCP and Megaco is because those are device 
     > control protocols intended for comand and control of an endpoint, as 
     > opposed to peer signaling protocols like SIP. 
     > 
     > --On Tuesday, February 04, 2003 12:24 PM +0200 Alex Agranov 
     > <sagranov@COMGATES.co.il> wrote: 
     > 
     > > 
     > > Hi, 
     > > 
     > > I tried to ask this question in sip-implementors maillist but nobody 
     > > could answer it there... 
     > > 
     > > What is the proper way to indicate Silence Suppression and Echo 
     > > Cancelation coder parameters in SDP? In MGCP these parameters are 
     > > passed in LCO element, but for SIP Offer/Answer model these 
     > parameters 
     > > should be passed in SDP. Right? 
     > > 
     > > There is RFC3108 that defines "ecan" and "silenceSupp" 
     > attributes, but it 
     > > applies to VoATM. Is it applicable to VoIP too? 
     > > 
     > > Some late drafts, e.g. 
     > draft-ietf-sipping-realtimefax-00.txt, seem to 
     > > assume  RFC3108 attributes are valid for VoIP. Is this correct? 
     > > 
     > > Best regards, 
     > >                 Alex Agranov 
     > > --- 
     > > Senior Software Engineer 
     > > COMGATES Ltd. 
     > > 15 Hagalim Avenue 
     > > Herzliya, 46725 
     > > Israel 
     > > Tel. +972.9.950.0404,  Ext: 228 
     > > Fax. +972.9.950.0385 
     > > Mobile. +972.54.928435 
     > 
     > ------------------------ 
     > David R. Oran 
     > Cisco Systems 
     > 7 Ladyslipper Lane 
     > Acton, MA 01720 
     > Office: +1 978 264 2048 
     > VoIP: +1 408 571 4576 
     > Email: oran@cisco.com 
     > 
==============================================================================
Qiru Zhou                                 (http://www.bell-labs.com/org/1133/)
Converged Network and Service Research  Bell Laboratories, Lucent Technologies
600 Mountain Avenue, 2D428, Murray Hill, NJ 07974, USA 
tel +1 908 582 4562  | fax +1 908 582 7308  |     qzhou@research.bell-labs.com
==============================================================================
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