Hadriel Kaplan | 2 Feb 22:49 2006

RE: Re: comedia-tls: proposed change for certificate identities

FWIW, I like this change a lot.   
-hadriel

> -----Original Message-----
> From: mmusic-bounces <at> ietf.org [mailto:mmusic-bounces <at> ietf.org] On Behalf
> Of Jonathan Lennox
> Sent: Monday, January 30, 2006 8:22 AM
> To: Sam Hartman
> Cc: mankin <at> psg.com; EKR; mmusic <at> ietf.org; jon.peterson <at> neustar.biz;
> tls <at> ietf.org
> Subject: [MMUSIC] Re: comedia-tls: proposed change for certificate
> identities
> 
> On Sunday, January 29 2006, "Sam Hartman" wrote to "Jonathan Lennox,
> mmusic <at> ietf.org, tls <at> ietf.org, EKR, jon.peterson <at> neustar.biz,
> mankin <at> psg.com" saying:
> 
> > >>>>> "Jonathan" == Jonathan Lennox <lennox <at> cs.columbia.edu> writes:
> >
> >
> >     Jonathan> * If your SDP was sent over an end-to-end authenticated
> >     Jonathan> and integrity-protected channel, there are no
> >     Jonathan> constraints on what identity you assert in your TLS
> >     Jonathan> certificates; you can use anything you want.  Identity
> >     Jonathan> is handled by the transporting protocol.
> >
> > What about hop-by-hop authenticated with each hop integrity protected.
> > IMHO in many environments, this presents acceptable man-in-the-middle
> > risk.  I think that it may present significantly easier to deal with
> > usability issues so I think it should at least be allowed if not
(Continue reading)

Burger, Eric | 3 Feb 03:33 2006

RE: RTSP & Megaco

Is the question “Why is SDP different for Megaco?” or is the question “Why can’t I use RTSP/SIP for Megaco?”

 

From: mmusic-bounces <at> ietf.org [mailto:mmusic-bounces <at> ietf.org] On Behalf Of Osher Hmelnizky
Sent: Sunday, January 08, 2006 3:10 AM
To: mmusic <at> ietf.org
Subject: [MMUSIC] RTSP & Megaco

 

Is it any draft developing regarding RTSP and Megaco.

All drafts speak on SIP & RTSP implementations. Why megaco can’t work in the same manner.

 

Osher

 

R&D Audiocodes

_______________________________________________
mmusic mailing list
mmusic <at> ietf.org
https://www1.ietf.org/mailman/listinfo/mmusic
canetti | 3 Feb 05:57 2006
Picon

Last Call for draft-ietf-msec-mikey-rsa-r-02.txt


Folks,

This is an MSEC last call on draft-ietf-msec-mikey-rsa-r-02.txt.
Version 02 was posted recently, and the document seems quite stable.
Please take time to read and comment.

MMUSIC folks: The main intended use of the proposed mode of MIKEY
is within mmusic protocols. Thus, although officially this is only an
MSEC LC, you may want to read and comment as well.

The LC will end in two weeks, on Friday Feb 17.

Thanks,

Ran

The latest version of the draft is available at:

http://www.ietf.org/internet-drafts/draft-ietf-msec-mikey-rsa-r-02.txt
Osher Hmelnizky | 5 Feb 08:41 2006

RE: Using RTSP with SIP

Hi,

 

What I’m trying to say is: why can’t I use SIP for the call establishment and RTSP for the stream control. How can synchronize two protocols on the MRF side?

On the client side it is no problem, but server side should be aware that two sessions are interconnected to the one specific call.

Such synchronization I saw in the SIP/MRCP draft (draft-ietf-speechsc-mrcpv2-09).

 

Thanks a lot

 

Osher

 

From: Burger, Eric [mailto:eburger <at> brooktrout.com]
Sent: Friday, February 03, 2006 4:34 AM
To: Osher Hmelnizky
Subject: RE: [MMUSIC] Using RTSP with SIP

 

What are you trying to accomplish?  Two sessions are two sessions.  It is either SIP or RTSP.  MRCP (not an IETF standard) has two, distinct sessions.

 

From: mmusic-bounces <at> ietf.org [mailto:mmusic-bounces <at> ietf.org] On Behalf Of Osher Hmelnizky
Sent: Tuesday, January 10, 2006 7:26 AM
To: mmusic <at> ietf.org
Subject: [MMUSIC] Using RTSP with SIP

 

In the last MRCP draft (draft-ietf-speechsc-mrcpv2-09), in order to synchronize the SIP and MRCP Sessios, defined

a=channel:xxx

 

as part of application SDP.

 

I’m trying to find out how I can synchronize SIP and RTSP streams at the Media Server layer without any indication.

 

As a=channel defined at the SDP level can I use this parameter to sync between SIP and RTSP Streams.

 

Thanks a lot.

 

Osher

 

 

 

 

_______________________________________________
mmusic mailing list
mmusic <at> ietf.org
https://www1.ietf.org/mailman/listinfo/mmusic
Burger, Eric | 5 Feb 14:17 2006

RE: Using RTSP with SIP

That is something totally different and explicitly outside the scope of MRCPv2.  What you want is something
like the mediactrl work (not chartered).  Search the I-D directory for mediactrl.  To subscribe, send
'subscribe mediactrl' to majordomo <at> lists.ubiquitysoftware.com

________________________________________
From: Osher Hmelnizky [mailto:Osher.Hmelnizky <at> audiocodes.com] 
Sent: Sunday, February 05, 2006 2:41 AM
To: Burger, Eric
Cc: mmusic <at> ietf.org
Subject: RE: [MMUSIC] Using RTSP with SIP

Hi,

What I'm trying to say is: why can't I use SIP for the call establishment and RTSP for the stream control. How
can synchronize two protocols on the MRF side? 
On the client side it is no problem, but server side should be aware that two sessions are interconnected to
the one specific call.
Such synchronization I saw in the SIP/MRCP draft (draft-ietf-speechsc-mrcpv2-09).

Thanks a lot

Osher

________________________________________
From: Burger, Eric [mailto:eburger <at> brooktrout.com] 
Sent: Friday, February 03, 2006 4:34 AM
To: Osher Hmelnizky
Subject: RE: [MMUSIC] Using RTSP with SIP

What are you trying to accomplish?  Two sessions are two sessions.  It is either SIP or RTSP.  MRCP (not
an IETF standard) has two, distinct sessions.

________________________________________
From: mmusic-bounces <at> ietf.org [mailto:mmusic-bounces <at> ietf.org] On Behalf Of Osher Hmelnizky
Sent: Tuesday, January 10, 2006 7:26 AM
To: mmusic <at> ietf.org
Subject: [MMUSIC] Using RTSP with SIP

In the last MRCP draft (draft-ietf-speechsc-mrcpv2-09), in order to synchronize the SIP and MRCP
Sessios, defined
       a=channel:xxx

as part of application SDP.

I'm trying to find out how I can synchronize SIP and RTSP streams at the Media Server layer without any indication.

As a=channel defined at the SDP level can I use this parameter to sync between SIP and RTSP Streams.

Thanks a lot.

Osher
Teodora Guenkova-Luy | 6 Feb 10:48 2006
Picon

Re: mmusic Digest, Vol 22, Issue 3

Hi,
SIP and RTSP are two different sessions. Their frameworks do not exclude 
the scenario that both of them shall be synchronised, however, there is 
no existing solution on such synchronisation. You can synchronize SIP 
and RTSP only at presentation level (i.e. SDP), like done for SIP and 
MRCP, as at session level SIP and RTSP are different instances that are 
not aware of each other. You could "misuse" the SDP SIP/MRCP attribute 
"a=channel:xxx" with your own parameters for RTSP, or use the SDP "i=" 
line also with your own paramentes, or define your own attribute line, 
e.g. "a=stacksynch:whatever". However, all these are non-standard solutions.
KR,
Teodora

mmusic-request <at> ietf.org wrote:

>
>Message: 1
>Date: Sun, 5 Feb 2006 09:41:06 +0200
>From: "Osher Hmelnizky" <Osher.Hmelnizky <at> audiocodes.com>
>Subject: RE: [MMUSIC] Using RTSP with SIP
>To: "Burger, Eric" <eburger <at> brooktrout.com>
>Cc: mmusic <at> ietf.org
>Message-ID:
>	<79B4F738DDD4EF4F85A4641A0FE5EFD601C89990 <at> aclmsg.corp.audiocodes.com>
>Content-Type: text/plain; charset="us-ascii"
>
>Hi,
>
> 
>
>What I'm trying to say is: why can't I use SIP for the call
>establishment and RTSP for the stream control. How can synchronize two
>protocols on the MRF side? 
>
>On the client side it is no problem, but server side should be aware
>that two sessions are interconnected to the one specific call.
>
>Such synchronization I saw in the SIP/MRCP draft
>(draft-ietf-speechsc-mrcpv2-09).
>
> 
>
>Thanks a lot
>
> 
>
>Osher
>
> 
>
>________________________________
>
>From: Burger, Eric [mailto:eburger <at> brooktrout.com] 
>Sent: Friday, February 03, 2006 4:34 AM
>To: Osher Hmelnizky
>Subject: RE: [MMUSIC] Using RTSP with SIP
>
> 
>
>What are you trying to accomplish?  Two sessions are two sessions.  It
>is either SIP or RTSP.  MRCP (not an IETF standard) has two, distinct
>sessions.
>
> 
>
>________________________________
>
>From: mmusic-bounces <at> ietf.org [mailto:mmusic-bounces <at> ietf.org] On Behalf
>Of Osher Hmelnizky
>Sent: Tuesday, January 10, 2006 7:26 AM
>To: mmusic <at> ietf.org
>Subject: [MMUSIC] Using RTSP with SIP
>
> 
>
>In the last MRCP draft (draft-ietf-speechsc-mrcpv2-09), in order to
>synchronize the SIP and MRCP Sessios, defined
>
>a=channel:xxx
>
> 
>
>as part of application SDP.
>
> 
>
>I'm trying to find out how I can synchronize SIP and RTSP streams at the
>Media Server layer without any indication.
>
> 
>
>As a=channel defined at the SDP level can I use this parameter to sync
>between SIP and RTSP Streams.
>
> 
>
>Thanks a lot.
>
> 
>
>Osher
>
> 
>
> 
>
> 
>
> 
>
>-------------- next part --------------
>An HTML attachment was scrubbed...
>URL: http://www1.ietf.org/pipermail/mmusic/attachments/20060205/0ce380c0/attachment.htm
>
>------------------------------
>
>Message: 2
>Date: Sun, 5 Feb 2006 08:17:22 -0500
>From: "Burger, Eric" <eburger <at> brooktrout.com>
>Subject: RE: [MMUSIC] Using RTSP with SIP
>To: "Osher Hmelnizky" <Osher.Hmelnizky <at> audiocodes.com>
>Cc: mmusic <at> ietf.org
>Message-ID:
>	<330A23D8336C0346B5C1A5BB196666470228F906 <at> ATLANTIS.Brooktrout.com>
>Content-Type: text/plain;	charset="iso-8859-1"
>
>That is something totally different and explicitly outside the scope of MRCPv2.  What you want is
something like the mediactrl work (not chartered).  Search the I-D directory for mediactrl.  To
subscribe, send 'subscribe mediactrl' to majordomo <at> lists.ubiquitysoftware.com
>
>________________________________________
>From: Osher Hmelnizky [mailto:Osher.Hmelnizky <at> audiocodes.com] 
>Sent: Sunday, February 05, 2006 2:41 AM
>To: Burger, Eric
>Cc: mmusic <at> ietf.org
>Subject: RE: [MMUSIC] Using RTSP with SIP
>
>Hi,
>
>What I'm trying to say is: why can't I use SIP for the call establishment and RTSP for the stream control. How
can synchronize two protocols on the MRF side? 
>On the client side it is no problem, but server side should be aware that two sessions are interconnected to
the one specific call.
>Such synchronization I saw in the SIP/MRCP draft (draft-ietf-speechsc-mrcpv2-09).
>
>Thanks a lot
>
>Osher
>
>________________________________________
>From: Burger, Eric [mailto:eburger <at> brooktrout.com] 
>Sent: Friday, February 03, 2006 4:34 AM
>To: Osher Hmelnizky
>Subject: RE: [MMUSIC] Using RTSP with SIP
>
>What are you trying to accomplish?  Two sessions are two sessions.  It is either SIP or RTSP.  MRCP (not an IETF
standard) has two, distinct sessions.
>
>________________________________________
>From: mmusic-bounces <at> ietf.org [mailto:mmusic-bounces <at> ietf.org] On Behalf Of Osher Hmelnizky
>Sent: Tuesday, January 10, 2006 7:26 AM
>To: mmusic <at> ietf.org
>Subject: [MMUSIC] Using RTSP with SIP
>
>In the last MRCP draft (draft-ietf-speechsc-mrcpv2-09), in order to synchronize the SIP and MRCP
Sessios, defined
>       a=channel:xxx
>       
>as part of application SDP.
>
>I'm trying to find out how I can synchronize SIP and RTSP streams at the Media Server layer without any indication.
>
>As a=channel defined at the SDP level can I use this parameter to sync between SIP and RTSP Streams.
>
>Thanks a lot.
>
>Osher
>
>
>
>
>
>
>------------------------------
>
>_______________________________________________
>mmusic mailing list
>mmusic <at> ietf.org
>https://www1.ietf.org/mailman/listinfo/mmusic
>
>
>End of mmusic Digest, Vol 22, Issue 3
>*************************************
>  
>

--

-- 
-----------------------------------------
Teodora Guenkova-Luy

Dept. Distributed Systems, University of Ulm, 
Oberer Eselsberg, 89069 Ulm, Germany

Tel.: +49 (0731) 502-4148 
FAX: +49 (0731) 502-4142

e-Mail: teodora.guenkova-luy <at> uni-ulm.de

Home Page: http://www-vs.informatik.uni-ulm.de/~Guenkova/ 

-----------------------------------------
Magnus Westerlund | 6 Feb 10:46 2006
Picon

Re: [Rmt] SDP descriptors for ALC

Hi Imed and Rod,

I do have some comments on your proposal.

First of all why have both a=flute-tsi and an a=alc-tsi. Wouldn't a 
a=tsi attribute be more appropriate. To my understanding there is no 
difference in TSI between the two. Also if some other fully specified 
protocol uses TSI they can reuse the TSI attribute. As the protocol will 
identify if it is flute or ALC, or something else in the future I don't 
see a risk of misinterpretation. The same comment applies also to the 
channel attribute. However here something more than a=ch is probably a 
good thing to indicate that it is the number of transport channels 
rather than related to media. Maybe a=trn-ch?

I am also concerned by the use of * and no rules for the ALC/UDP 
protocol. Having something that is underspecified but no defined rules 
for how fully specified mechanisms are identified seem to be a bad 
choice. I would recommend that one uses some type of identifier space 
here. I don't think media types are the correct one, but some space 
should be set up to identify applications.

cheers

Magnus

--

-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund <at> ericsson.com
Colin Perkins | 6 Feb 10:56 2006

Re: Re: mmusic Digest, Vol 22, Issue 3

SDP has the concept of "m=control ..." to signal other control  
sessions. If one were to define how to signal RTSP sessions could be  
setup using SIP, that would likely be the appropriate way to do it.

This subject comes up regularly - maybe it would be appropriate for  
someone to write a draft fleshing out the details?

Colin

On 6 Feb 2006, at 09:48, Teodora Guenkova-Luy wrote:
> SIP and RTSP are two different sessions. Their frameworks do not  
> exclude the scenario that both of them shall be synchronised,  
> however, there is no existing solution on such synchronisation. You  
> can synchronize SIP and RTSP only at presentation level (i.e. SDP),  
> like done for SIP and MRCP, as at session level SIP and RTSP are  
> different instances that are not aware of each other. You could  
> "misuse" the SDP SIP/MRCP attribute "a=channel:xxx" with your own  
> parameters for RTSP, or use the SDP "i=" line also with your own  
> paramentes, or define your own attribute line, e.g.  
> "a=stacksynch:whatever". However, all these are non-standard  
> solutions.
> KR,
> Teodora
>
> mmusic-request <at> ietf.org wrote:
>
>>
>> Message: 1
>> Date: Sun, 5 Feb 2006 09:41:06 +0200
>> From: "Osher Hmelnizky" <Osher.Hmelnizky <at> audiocodes.com>
>> Subject: RE: [MMUSIC] Using RTSP with SIP
>> To: "Burger, Eric" <eburger <at> brooktrout.com>
>> Cc: mmusic <at> ietf.org
>> Message-ID:
>> 	 
>> <79B4F738DDD4EF4F85A4641A0FE5EFD601C89990 <at> aclmsg.corp.audiocodes.com>
>> Content-Type: text/plain; charset="us-ascii"
>>
>> Hi,
>>
>>
>> What I'm trying to say is: why can't I use SIP for the call
>> establishment and RTSP for the stream control. How can synchronize  
>> two
>> protocols on the MRF side?
>> On the client side it is no problem, but server side should be aware
>> that two sessions are interconnected to the one specific call.
>>
>> Such synchronization I saw in the SIP/MRCP draft
>> (draft-ietf-speechsc-mrcpv2-09).
>>
>>
>> Thanks a lot
>>
>>
>> Osher
>>
>>
>> ________________________________
>>
>> From: Burger, Eric [mailto:eburger <at> brooktrout.com] Sent: Friday,  
>> February 03, 2006 4:34 AM
>> To: Osher Hmelnizky
>> Subject: RE: [MMUSIC] Using RTSP with SIP
>>
>>
>> What are you trying to accomplish?  Two sessions are two  
>> sessions.  It
>> is either SIP or RTSP.  MRCP (not an IETF standard) has two, distinct
>> sessions.
>>
>>
>> ________________________________
>>
>> From: mmusic-bounces <at> ietf.org [mailto:mmusic-bounces <at> ietf.org] On  
>> Behalf
>> Of Osher Hmelnizky
>> Sent: Tuesday, January 10, 2006 7:26 AM
>> To: mmusic <at> ietf.org
>> Subject: [MMUSIC] Using RTSP with SIP
>>
>>
>> In the last MRCP draft (draft-ietf-speechsc-mrcpv2-09), in order to
>> synchronize the SIP and MRCP Sessios, defined
>>
>> a=channel:xxx
>>
>>
>> as part of application SDP.
>>
>>
>> I'm trying to find out how I can synchronize SIP and RTSP streams  
>> at the
>> Media Server layer without any indication.
>>
>>
>> As a=channel defined at the SDP level can I use this parameter to  
>> sync
>> between SIP and RTSP Streams.
>>
>>
>> Thanks a lot.
>>
>>
>> Osher
>>
>>
>>
>>
>>
>> -------------- next part --------------
>> An HTML attachment was scrubbed...
>> URL: http://www1.ietf.org/pipermail/mmusic/attachments/ 
>> 20060205/0ce380c0/attachment.htm
>>
>> ------------------------------
>>
>> Message: 2
>> Date: Sun, 5 Feb 2006 08:17:22 -0500
>> From: "Burger, Eric" <eburger <at> brooktrout.com>
>> Subject: RE: [MMUSIC] Using RTSP with SIP
>> To: "Osher Hmelnizky" <Osher.Hmelnizky <at> audiocodes.com>
>> Cc: mmusic <at> ietf.org
>> Message-ID:
>> 	<330A23D8336C0346B5C1A5BB196666470228F906 <at> ATLANTIS.Brooktrout.com>
>> Content-Type: text/plain;	charset="iso-8859-1"
>>
>> That is something totally different and explicitly outside the  
>> scope of MRCPv2.  What you want is something like the mediactrl  
>> work (not chartered).  Search the I-D directory for mediactrl.  To  
>> subscribe, send 'subscribe mediactrl' to  
>> majordomo <at> lists.ubiquitysoftware.com
>>
>> ________________________________________
>> From: Osher Hmelnizky [mailto:Osher.Hmelnizky <at> audiocodes.com]  
>> Sent: Sunday, February 05, 2006 2:41 AM
>> To: Burger, Eric
>> Cc: mmusic <at> ietf.org
>> Subject: RE: [MMUSIC] Using RTSP with SIP
>>
>> Hi,
>>
>> What I'm trying to say is: why can't I use SIP for the call  
>> establishment and RTSP for the stream control. How can synchronize  
>> two protocols on the MRF side? On the client side it is no  
>> problem, but server side should be aware that two sessions are  
>> interconnected to the one specific call.
>> Such synchronization I saw in the SIP/MRCP draft (draft-ietf- 
>> speechsc-mrcpv2-09).
>>
>> Thanks a lot
>>
>> Osher
>>
>> ________________________________________
>> From: Burger, Eric [mailto:eburger <at> brooktrout.com] Sent: Friday,  
>> February 03, 2006 4:34 AM
>> To: Osher Hmelnizky
>> Subject: RE: [MMUSIC] Using RTSP with SIP
>>
>> What are you trying to accomplish?  Two sessions are two  
>> sessions.  It is either SIP or RTSP.  MRCP (not an IETF standard)  
>> has two, distinct sessions.
>>
>> ________________________________________
>> From: mmusic-bounces <at> ietf.org [mailto:mmusic-bounces <at> ietf.org] On  
>> Behalf Of Osher Hmelnizky
>> Sent: Tuesday, January 10, 2006 7:26 AM
>> To: mmusic <at> ietf.org
>> Subject: [MMUSIC] Using RTSP with SIP
>>
>> In the last MRCP draft (draft-ietf-speechsc-mrcpv2-09), in order  
>> to synchronize the SIP and MRCP Sessios, defined
>>       a=channel:xxx
>>       as part of application SDP.
>>
>> I'm trying to find out how I can synchronize SIP and RTSP streams  
>> at the Media Server layer without any indication.
>>
>> As a=channel defined at the SDP level can I use this parameter to  
>> sync between SIP and RTSP Streams.
>>
>> Thanks a lot.
>>
>> Osher
>>
>>
>>
>>
>>
>>
>> ------------------------------
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic <at> ietf.org
>> https://www1.ietf.org/mailman/listinfo/mmusic
>>
>>
>> End of mmusic Digest, Vol 22, Issue 3
>> *************************************
>>
>
> -- 
> -----------------------------------------
> Teodora Guenkova-Luy
>
> Dept. Distributed Systems, University of Ulm, Oberer Eselsberg,  
> 89069 Ulm, Germany
>
> Tel.: +49 (0731) 502-4148 FAX: +49 (0731) 502-4142
>
> e-Mail: teodora.guenkova-luy <at> uni-ulm.de
>
> Home Page: http://www-vs.informatik.uni-ulm.de/~Guenkova/
> -----------------------------------------
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic <at> ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
Robert Hagens | 6 Feb 16:02 2006

RE: Re: mmusic Digest, Vol 22, Issue 3

The other non-standard aspect of SIP/RTSP is that there is a good chuck of
overlap between SIP and RTSP. If you (for example) negotiate the SDP with
SIP, you really don't need to bother with the DESCRIBE and SETUP of RTSP.
After the SIP negotiation, you're ready to PLAY. This leads one to question
if all of the complexity of RTSP is required in these cases; a simpler
version of media control being desirable.

Rob

-----Original Message-----
From: mmusic-bounces <at> ietf.org [mailto:mmusic-bounces <at> ietf.org] On Behalf Of
Colin Perkins
Sent: Monday, February 06, 2006 2:57 AM
To: Teodora Guenkova-Luy
Cc: mmusic <at> ietf.org; Osher.Hmelnizky <at> audiocodes.com
Subject: Re: [MMUSIC] Re: mmusic Digest, Vol 22, Issue 3

SDP has the concept of "m=control ..." to signal other control  
sessions. If one were to define how to signal RTSP sessions could be  
setup using SIP, that would likely be the appropriate way to do it.

This subject comes up regularly - maybe it would be appropriate for  
someone to write a draft fleshing out the details?

Colin

On 6 Feb 2006, at 09:48, Teodora Guenkova-Luy wrote:
> SIP and RTSP are two different sessions. Their frameworks do not  
> exclude the scenario that both of them shall be synchronised,  
> however, there is no existing solution on such synchronisation. You  
> can synchronize SIP and RTSP only at presentation level (i.e. SDP),  
> like done for SIP and MRCP, as at session level SIP and RTSP are  
> different instances that are not aware of each other. You could  
> "misuse" the SDP SIP/MRCP attribute "a=channel:xxx" with your own  
> parameters for RTSP, or use the SDP "i=" line also with your own  
> paramentes, or define your own attribute line, e.g.  
> "a=stacksynch:whatever". However, all these are non-standard  
> solutions.
> KR,
> Teodora
>
> mmusic-request <at> ietf.org wrote:
>
>>
>> Message: 1
>> Date: Sun, 5 Feb 2006 09:41:06 +0200
>> From: "Osher Hmelnizky" <Osher.Hmelnizky <at> audiocodes.com>
>> Subject: RE: [MMUSIC] Using RTSP with SIP
>> To: "Burger, Eric" <eburger <at> brooktrout.com>
>> Cc: mmusic <at> ietf.org
>> Message-ID:
>> 	 
>> <79B4F738DDD4EF4F85A4641A0FE5EFD601C89990 <at> aclmsg.corp.audiocodes.com>
>> Content-Type: text/plain; charset="us-ascii"
>>
>> Hi,
>>
>>
>> What I'm trying to say is: why can't I use SIP for the call
>> establishment and RTSP for the stream control. How can synchronize  
>> two
>> protocols on the MRF side?
>> On the client side it is no problem, but server side should be aware
>> that two sessions are interconnected to the one specific call.
>>
>> Such synchronization I saw in the SIP/MRCP draft
>> (draft-ietf-speechsc-mrcpv2-09).
>>
>>
>> Thanks a lot
>>
>>
>> Osher
>>
>>
>> ________________________________
>>
>> From: Burger, Eric [mailto:eburger <at> brooktrout.com] Sent: Friday,  
>> February 03, 2006 4:34 AM
>> To: Osher Hmelnizky
>> Subject: RE: [MMUSIC] Using RTSP with SIP
>>
>>
>> What are you trying to accomplish?  Two sessions are two  
>> sessions.  It
>> is either SIP or RTSP.  MRCP (not an IETF standard) has two, distinct
>> sessions.
>>
>>
>> ________________________________
>>
>> From: mmusic-bounces <at> ietf.org [mailto:mmusic-bounces <at> ietf.org] On  
>> Behalf
>> Of Osher Hmelnizky
>> Sent: Tuesday, January 10, 2006 7:26 AM
>> To: mmusic <at> ietf.org
>> Subject: [MMUSIC] Using RTSP with SIP
>>
>>
>> In the last MRCP draft (draft-ietf-speechsc-mrcpv2-09), in order to
>> synchronize the SIP and MRCP Sessios, defined
>>
>> a=channel:xxx
>>
>>
>> as part of application SDP.
>>
>>
>> I'm trying to find out how I can synchronize SIP and RTSP streams  
>> at the
>> Media Server layer without any indication.
>>
>>
>> As a=channel defined at the SDP level can I use this parameter to  
>> sync
>> between SIP and RTSP Streams.
>>
>>
>> Thanks a lot.
>>
>>
>> Osher
>>
>>
>>
>>
>>
>> -------------- next part --------------
>> An HTML attachment was scrubbed...
>> URL: http://www1.ietf.org/pipermail/mmusic/attachments/ 
>> 20060205/0ce380c0/attachment.htm
>>
>> ------------------------------
>>
>> Message: 2
>> Date: Sun, 5 Feb 2006 08:17:22 -0500
>> From: "Burger, Eric" <eburger <at> brooktrout.com>
>> Subject: RE: [MMUSIC] Using RTSP with SIP
>> To: "Osher Hmelnizky" <Osher.Hmelnizky <at> audiocodes.com>
>> Cc: mmusic <at> ietf.org
>> Message-ID:
>> 	<330A23D8336C0346B5C1A5BB196666470228F906 <at> ATLANTIS.Brooktrout.com>
>> Content-Type: text/plain;	charset="iso-8859-1"
>>
>> That is something totally different and explicitly outside the  
>> scope of MRCPv2.  What you want is something like the mediactrl  
>> work (not chartered).  Search the I-D directory for mediactrl.  To  
>> subscribe, send 'subscribe mediactrl' to  
>> majordomo <at> lists.ubiquitysoftware.com
>>
>> ________________________________________
>> From: Osher Hmelnizky [mailto:Osher.Hmelnizky <at> audiocodes.com]  
>> Sent: Sunday, February 05, 2006 2:41 AM
>> To: Burger, Eric
>> Cc: mmusic <at> ietf.org
>> Subject: RE: [MMUSIC] Using RTSP with SIP
>>
>> Hi,
>>
>> What I'm trying to say is: why can't I use SIP for the call  
>> establishment and RTSP for the stream control. How can synchronize  
>> two protocols on the MRF side? On the client side it is no  
>> problem, but server side should be aware that two sessions are  
>> interconnected to the one specific call.
>> Such synchronization I saw in the SIP/MRCP draft (draft-ietf- 
>> speechsc-mrcpv2-09).
>>
>> Thanks a lot
>>
>> Osher
>>
>> ________________________________________
>> From: Burger, Eric [mailto:eburger <at> brooktrout.com] Sent: Friday,  
>> February 03, 2006 4:34 AM
>> To: Osher Hmelnizky
>> Subject: RE: [MMUSIC] Using RTSP with SIP
>>
>> What are you trying to accomplish?  Two sessions are two  
>> sessions.  It is either SIP or RTSP.  MRCP (not an IETF standard)  
>> has two, distinct sessions.
>>
>> ________________________________________
>> From: mmusic-bounces <at> ietf.org [mailto:mmusic-bounces <at> ietf.org] On  
>> Behalf Of Osher Hmelnizky
>> Sent: Tuesday, January 10, 2006 7:26 AM
>> To: mmusic <at> ietf.org
>> Subject: [MMUSIC] Using RTSP with SIP
>>
>> In the last MRCP draft (draft-ietf-speechsc-mrcpv2-09), in order  
>> to synchronize the SIP and MRCP Sessios, defined
>>       a=channel:xxx
>>       as part of application SDP.
>>
>> I'm trying to find out how I can synchronize SIP and RTSP streams  
>> at the Media Server layer without any indication.
>>
>> As a=channel defined at the SDP level can I use this parameter to  
>> sync between SIP and RTSP Streams.
>>
>> Thanks a lot.
>>
>> Osher
>>
>>
>>
>>
>>
>>
>> ------------------------------
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic <at> ietf.org
>> https://www1.ietf.org/mailman/listinfo/mmusic
>>
>>
>> End of mmusic Digest, Vol 22, Issue 3
>> *************************************
>>
>
> -- 
> -----------------------------------------
> Teodora Guenkova-Luy
>
> Dept. Distributed Systems, University of Ulm, Oberer Eselsberg,  
> 89069 Ulm, Germany
>
> Tel.: +49 (0731) 502-4148 FAX: +49 (0731) 502-4142
>
> e-Mail: teodora.guenkova-luy <at> uni-ulm.de
>
> Home Page: http://www-vs.informatik.uni-ulm.de/~Guenkova/
> -----------------------------------------
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic <at> ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

_______________________________________________
mmusic mailing list
mmusic <at> ietf.org
https://www1.ietf.org/mailman/listinfo/mmusic
Osher Hmelnizky | 7 Feb 08:59 2006

RE: Re: mmusic Digest, Vol 22, Issue 3

Hi,

When I'm thinking on the Media server implementation in the IMS
architecture, it's clear that most of the SIP devices running over the
UDP for call establishment. Most of the xml-types streaming/media
controls over the SIP using INFO approaches. I'm not sure that this is a
right approach for real time streaming control (explicitly for video).
RTSP2 (the last draft) defines RTSP to use TCP as transport protocol.
TCP is the right approach for real time streaming control.

I adopt SIP/Info streaming control for announcement and recording, but
for real time stream control I think it's not enough, because of the big
delays when, for example Application Server should be involved in the
DTMF detection and creation of the appropriate control to the media
server.

SIP/MRCP2&RTSP seems me right approach for media server control. SIP for
session establishment and MRCP or RTSP for the streaming control and
media services.
I think it should be one common technique for synchronization between
SIP and other protocol like RTSP. It may be done using the SDP(for
example SIP/MRCP2).

Any thoughts?

Kind regards

Osher

-----Original Message-----
From: Robert Hagens [mailto:rhagens <at> envysion.com] 
Sent: Monday, February 06, 2006 5:03 PM
To: 'Colin Perkins'; 'Teodora Guenkova-Luy'
Cc: mmusic <at> ietf.org; Osher Hmelnizky
Subject: RE: [MMUSIC] Re: mmusic Digest, Vol 22, Issue 3

The other non-standard aspect of SIP/RTSP is that there is a good chuck
of
overlap between SIP and RTSP. If you (for example) negotiate the SDP
with
SIP, you really don't need to bother with the DESCRIBE and SETUP of
RTSP.
After the SIP negotiation, you're ready to PLAY. This leads one to
question
if all of the complexity of RTSP is required in these cases; a simpler
version of media control being desirable.

Rob

-----Original Message-----
From: mmusic-bounces <at> ietf.org [mailto:mmusic-bounces <at> ietf.org] On Behalf
Of
Colin Perkins
Sent: Monday, February 06, 2006 2:57 AM
To: Teodora Guenkova-Luy
Cc: mmusic <at> ietf.org; Osher.Hmelnizky <at> audiocodes.com
Subject: Re: [MMUSIC] Re: mmusic Digest, Vol 22, Issue 3

SDP has the concept of "m=control ..." to signal other control  
sessions. If one were to define how to signal RTSP sessions could be  
setup using SIP, that would likely be the appropriate way to do it.

This subject comes up regularly - maybe it would be appropriate for  
someone to write a draft fleshing out the details?

Colin

On 6 Feb 2006, at 09:48, Teodora Guenkova-Luy wrote:
> SIP and RTSP are two different sessions. Their frameworks do not  
> exclude the scenario that both of them shall be synchronised,  
> however, there is no existing solution on such synchronisation. You  
> can synchronize SIP and RTSP only at presentation level (i.e. SDP),  
> like done for SIP and MRCP, as at session level SIP and RTSP are  
> different instances that are not aware of each other. You could  
> "misuse" the SDP SIP/MRCP attribute "a=channel:xxx" with your own  
> parameters for RTSP, or use the SDP "i=" line also with your own  
> paramentes, or define your own attribute line, e.g.  
> "a=stacksynch:whatever". However, all these are non-standard  
> solutions.
> KR,
> Teodora
>
> mmusic-request <at> ietf.org wrote:
>
>>
>> Message: 1
>> Date: Sun, 5 Feb 2006 09:41:06 +0200
>> From: "Osher Hmelnizky" <Osher.Hmelnizky <at> audiocodes.com>
>> Subject: RE: [MMUSIC] Using RTSP with SIP
>> To: "Burger, Eric" <eburger <at> brooktrout.com>
>> Cc: mmusic <at> ietf.org
>> Message-ID:
>> 	 
>> <79B4F738DDD4EF4F85A4641A0FE5EFD601C89990 <at> aclmsg.corp.audiocodes.com>
>> Content-Type: text/plain; charset="us-ascii"
>>
>> Hi,
>>
>>
>> What I'm trying to say is: why can't I use SIP for the call
>> establishment and RTSP for the stream control. How can synchronize  
>> two
>> protocols on the MRF side?
>> On the client side it is no problem, but server side should be aware
>> that two sessions are interconnected to the one specific call.
>>
>> Such synchronization I saw in the SIP/MRCP draft
>> (draft-ietf-speechsc-mrcpv2-09).
>>
>>
>> Thanks a lot
>>
>>
>> Osher
>>
>>
>> ________________________________
>>
>> From: Burger, Eric [mailto:eburger <at> brooktrout.com] Sent: Friday,  
>> February 03, 2006 4:34 AM
>> To: Osher Hmelnizky
>> Subject: RE: [MMUSIC] Using RTSP with SIP
>>
>>
>> What are you trying to accomplish?  Two sessions are two  
>> sessions.  It
>> is either SIP or RTSP.  MRCP (not an IETF standard) has two, distinct
>> sessions.
>>
>>
>> ________________________________
>>
>> From: mmusic-bounces <at> ietf.org [mailto:mmusic-bounces <at> ietf.org] On  
>> Behalf
>> Of Osher Hmelnizky
>> Sent: Tuesday, January 10, 2006 7:26 AM
>> To: mmusic <at> ietf.org
>> Subject: [MMUSIC] Using RTSP with SIP
>>
>>
>> In the last MRCP draft (draft-ietf-speechsc-mrcpv2-09), in order to
>> synchronize the SIP and MRCP Sessios, defined
>>
>> a=channel:xxx
>>
>>
>> as part of application SDP.
>>
>>
>> I'm trying to find out how I can synchronize SIP and RTSP streams  
>> at the
>> Media Server layer without any indication.
>>
>>
>> As a=channel defined at the SDP level can I use this parameter to  
>> sync
>> between SIP and RTSP Streams.
>>
>>
>> Thanks a lot.
>>
>>
>> Osher
>>
>>
>>
>>
>>
>> -------------- next part --------------
>> An HTML attachment was scrubbed...
>> URL: http://www1.ietf.org/pipermail/mmusic/attachments/ 
>> 20060205/0ce380c0/attachment.htm
>>
>> ------------------------------
>>
>> Message: 2
>> Date: Sun, 5 Feb 2006 08:17:22 -0500
>> From: "Burger, Eric" <eburger <at> brooktrout.com>
>> Subject: RE: [MMUSIC] Using RTSP with SIP
>> To: "Osher Hmelnizky" <Osher.Hmelnizky <at> audiocodes.com>
>> Cc: mmusic <at> ietf.org
>> Message-ID:
>>
<330A23D8336C0346B5C1A5BB196666470228F906 <at> ATLANTIS.Brooktrout.com>
>> Content-Type: text/plain;	charset="iso-8859-1"
>>
>> That is something totally different and explicitly outside the  
>> scope of MRCPv2.  What you want is something like the mediactrl  
>> work (not chartered).  Search the I-D directory for mediactrl.  To  
>> subscribe, send 'subscribe mediactrl' to  
>> majordomo <at> lists.ubiquitysoftware.com
>>
>> ________________________________________
>> From: Osher Hmelnizky [mailto:Osher.Hmelnizky <at> audiocodes.com]  
>> Sent: Sunday, February 05, 2006 2:41 AM
>> To: Burger, Eric
>> Cc: mmusic <at> ietf.org
>> Subject: RE: [MMUSIC] Using RTSP with SIP
>>
>> Hi,
>>
>> What I'm trying to say is: why can't I use SIP for the call  
>> establishment and RTSP for the stream control. How can synchronize  
>> two protocols on the MRF side? On the client side it is no  
>> problem, but server side should be aware that two sessions are  
>> interconnected to the one specific call.
>> Such synchronization I saw in the SIP/MRCP draft (draft-ietf- 
>> speechsc-mrcpv2-09).
>>
>> Thanks a lot
>>
>> Osher
>>
>> ________________________________________
>> From: Burger, Eric [mailto:eburger <at> brooktrout.com] Sent: Friday,  
>> February 03, 2006 4:34 AM
>> To: Osher Hmelnizky
>> Subject: RE: [MMUSIC] Using RTSP with SIP
>>
>> What are you trying to accomplish?  Two sessions are two  
>> sessions.  It is either SIP or RTSP.  MRCP (not an IETF standard)  
>> has two, distinct sessions.
>>
>> ________________________________________
>> From: mmusic-bounces <at> ietf.org [mailto:mmusic-bounces <at> ietf.org] On  
>> Behalf Of Osher Hmelnizky
>> Sent: Tuesday, January 10, 2006 7:26 AM
>> To: mmusic <at> ietf.org
>> Subject: [MMUSIC] Using RTSP with SIP
>>
>> In the last MRCP draft (draft-ietf-speechsc-mrcpv2-09), in order  
>> to synchronize the SIP and MRCP Sessios, defined
>>       a=channel:xxx
>>       as part of application SDP.
>>
>> I'm trying to find out how I can synchronize SIP and RTSP streams  
>> at the Media Server layer without any indication.
>>
>> As a=channel defined at the SDP level can I use this parameter to  
>> sync between SIP and RTSP Streams.
>>
>> Thanks a lot.
>>
>> Osher
>>
>>
>>
>>
>>
>>
>> ------------------------------
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic <at> ietf.org
>> https://www1.ietf.org/mailman/listinfo/mmusic
>>
>>
>> End of mmusic Digest, Vol 22, Issue 3
>> *************************************
>>
>
> -- 
> -----------------------------------------
> Teodora Guenkova-Luy
>
> Dept. Distributed Systems, University of Ulm, Oberer Eselsberg,  
> 89069 Ulm, Germany
>
> Tel.: +49 (0731) 502-4148 FAX: +49 (0731) 502-4142
>
> e-Mail: teodora.guenkova-luy <at> uni-ulm.de
>
> Home Page: http://www-vs.informatik.uni-ulm.de/~Guenkova/
> -----------------------------------------
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic <at> ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

_______________________________________________
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mmusic <at> ietf.org
https://www1.ietf.org/mailman/listinfo/mmusic

Gmane