Hi,
Talking about G711 fax fallback - I agree that for the direct call between 2 SIP UA's silence
suppression may be detected by each SIP UA when it recognizes the fax tone.
However let's look at the following scenario:
(SIP UA) ----SIP---- (PSTN GW) ----SS7---- (PSTN GW) ----SIP---- (SIP UA)
There is a need to pass information about silence mode between PSTN GW's,
so that they make proper connections in the corresponding MGW's.
For the regular SS7 call this information is included in SS7 message. But if we
have regular SIP UA's (not SIP-T GW's), there is no indication of silence suppression mode
in the SIP message. Hence there will be no indication of silence mode in SS7 message too.
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
> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Wednesday, February 05, 2003 1:32 AM
> To: Steve Silverman
> Cc: Alex Agranov; 'David R. Oran'; sip@ietf.org
> Subject: Re: [Sip] Silence Suppression in SDP for VoIP
>
>
> In some cases, the codec itself contains specific silence
> packets, and
> so their usage would be negotiated using the codec specific
> parameters.
> You can also use the specific payload format for comfort
> noise, and so
> support of that is negotiated using the SDP codec negotiation
> methodology in offer/answer. However, when silence
> suppression is done
> generally, by sending nothing, no signaling support is needed
> since all
> RTP receivers need to be prepared to receive nothing and treat it as
> silence. That is why there are both RTP timestamps and
> sequence numbers.
>
> In the case of fax, I still do not see the issue. The sending
> side would
> know that it is sending g.711 encoded fax, and therefore never detect
> silence.
>
> -Jonathan R.
>
> Steve Silverman wrote:
> > 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
> > >
> >
>
> --
> Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Ave.
> Chief Scientist First Floor
> dynamicsoft East Hanover, NJ 07936
> jdrosen@dynamicsoft.com FAX: (973) 952-5050
> http://www.jdrosen.net PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>