Re: [SIP] draft-camarillo-sip-sdp-00.txt
Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se> Mon, 16 October 2000 18:40 UTC
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Date: Mon, 16 Oct 2000 21:38:09 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
Organization: Ericsson
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
CC: "'Culpepper, Bert'" <bert.culpepper@intervoice-brite.com>, 'Michael Thomas' <mat@cisco.com>, 'Rohan Mahy' <rohan@cisco.com>, "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>, 'Henning Schulzrinne' <hgs@cs.columbia.edu>, 'mmusic' <confctrl@ISI.EDU>
Subject: Re: [SIP] draft-camarillo-sip-sdp-00.txt
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Hi Jonathan, Jonathan Rosenberg wrote: > > I just thought of an issue with this approach that is worth considering. > > The problem is that if A sends an INVITE to B, there cannot be more m lines > in the response from B that in the request from A. THis is needed to support > matching up of m lines. If A does not support the fid attribute, we are in today's situation (what we want to avioid), and B will have to do what you describe (re-INVITE right after the 200 OK). If A supports the fid attribute, it will include a fid attribute for each m line in the SDP. Upon reception of the INVITE, B knows that A understands the fid attribute because it appears in the SDP. Then, since both understand it, B can add as many m lines as it wishes, since the fid attribute will be used to align the media streams (rather than nth m lines matching). That's why the title of the draft is "SDP media alignment in SIP". So, this is a general mechanism not just for demultiplexing media flows but also for performing media alignment. Best regards, Gonzalo Now, lets say A is a PC, and it calls some user B > that needs to use one of these transcoders. In the 200 OK, B cannot add a > separate media stream for this codec. Rather, it will have to send a regular > 200 OK, and then immediately re-invite to add a media stream for that > specific codec, and remove that codec from the original media stream. Kind > of ugly, and possibly might get you glitches in the beginning of lost media. > > So, whilst I agree with the idea of what you are trying to do here (a > choose-one-of-M session feature), I am concerned about its general ability > to fit within the framework of SDP, which is widely recognized as limited > for such things. > > -Jonathan R. > > > -----Original Message----- > > From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@lmf.ericsson.se] > > Sent: Friday, October 13, 2000 4:30 AM > > To: Culpepper, Bert > > Cc: Michael Thomas; Rohan Mahy; Jonathan Rosenberg; > > sip@lists.bell-labs.com; Henning Schulzrinne; mmusic > > Subject: Re: [SIP] draft-camarillo-sip-sdp-00.txt > > > > > > Hi, > > > > I will try to summarize the discussions so far: > > > > The issue I am looking at is: "What happens when I need to receive a > > single media stream in different ports (or even different IP > > addresses) > > depending on the codec used at any moment". > > > > A media stream here is, for instance, the audio transmitted > > in a regular > > telephone call. > > > > With this issue in mind, it is good to know scenarios where this can > > happen and, of course, possible solutions. > > > > There are multiple scenarios where we can have this situation. > > > > -The device receiving voice is not the same as the device that handles > > DTMF tones. Thus, the media stream will have to be sent to the device > > handling voice when the RTP packets contain PCM voice (for > > instance) and > > to the device handling DTFM tones when they carry tones. > > > > -I need to use a transcoder because I have a cheap client > > that does not > > support all the audio codecs in the world. I want to receive PCM in my > > IP address, but I would like to receive GSM in the IP address of the > > transcoder. The transcoder will take care of sending the > > stream to my IP > > address once it has been transcoded. > > > > -Cellular terminals that have to configure radio bearers with specific > > codec information because some optimizations are applied. They want to > > receive media on different ports depending on the codec used. > > > > Folks have pointed out more scenarios where this might happen. > > > > The first solution I thought of was to send a single RTP stream to > > different ports (or different IP addresses). As the AVT guys pointed > > out, this would cause lots of problems. Mainly related to RTCP. > > Everybody agreed that a much better solution was to establish > > different > > RTP sessions. > > > > If different RTP sessions are established, this issue is > > transparent to > > RTP and RTCP. They are thus not affected. > > > > SDP backwards compatibility was also an issue, so a solution that does > > not break existing implementations has to be found. > > > > The fid (flow identifier) attribute I was describing in my > > previous mail > > would just be an identifier for the flow. That is, we might > > have several > > RTP streams, but all of them are related, because we are dealing with > > just one flow. > > > > So, an application receiving this parameter would know that all the m > > lines in the SDP with the same fid attribute (a=fid:12, for instance) > > are related. I believe this is useful information for the application. > > > > If an application does not understand the fid attribute, everything > > works properly. The only issue is that the application is not aware of > > this relation between m lines. > > > > > > Regarding Mike's question, this was already presented as > > input for SDPng > > in the last IETF. I am now after a solution for SDP. I agree > > that SDPng > > will solve all these issues. The quesion is: when? Everybody > > working on > > it is currently pretty busy. > > > > Best regards, > > > > Gonzalo > > > > > > > > "Culpepper, Bert" wrote: > > > > > > Hi: > > > > > > I've only followed this thread on the SIP list and read the recent > > > discussion on the AVT WG list, I've probably missed some > > related discussion > > > in other WGs. It appears to me, and as Gonzalo pointed > > out, separate RTP > > > streams are OK. The session description below describes > > two RTP streams > > > with only one active at any point in time (I believe the > > semantics of the > > > fid attribute shown indicate that I can choose to send only > > one) vs. a > > > single RTP stream with either of two encoding schemes. > > However, as a SIP UA > > > and having received that session description I know to send the two > > > encodings separately if I want to make the other endpoint > > happy. (And only > > > one at a time if I support the fid attribute.) It does > > seem reasonable that > > > it might complicate QoS, but I expect the benefit of the > > separate media > > > streams is justified when used. > > > > > > Anyway, segregating media streams (where exact > > synchronization isn't needed > > > and RTP is good enough) enables certain efficiencies when providing > > > telecommunications and like services in an IP network. > > This is only one > > > advantage I suspect. > > > > > > Regards, > > > Bert > > > > > > > -----Original Message----- > > > > From: Michael Thomas [mailto:mat@cisco.com] > > > > Sent: Thursday, October 12, 2000 10:22 AM > > > > To: Gonzalo Camarillo > > > > Cc: Culpepper, Bert; Rohan Mahy; Jonathan Rosenberg; > > > > sip@lists.bell-labs.com; Henning Schulzrinne; mmusic > > > > Subject: Re: [SIP] draft-camarillo-sip-sdp-00.txt > > > > > > > > > > > > Gonzalo Camarillo writes: > > > > > I would like my session description to look like: > > > > > > > > > > v=0 > > > > > o=John 289085535 289085535 IN IP4 first.example.com > > > > > t=0 0 > > > > > c=IN IP4 111.111.111.111 > > > > > m=audio 20000 RTP/AVP 0 > > > > > a=fid:1 > > > > > m=audio 20002 RTP/AVP 8 > > > > > a=fid:1 > > > > > > > > > > If an implementation does not understand the fid > > attribute it will try > > > > > to mix the audio streams. Since one is "empty" all the > > time, doing this > > > > > is not really a big deal. Thus, backwards > > compatibility is not a > > > > > concern. > > > > > > > > > > Do you have any concerns with this approach? > > > > > > > > The main problem I see is with the interaction with > > > > QoS reservations. Namely, that you'd expect the receiver > > > > to book reservations for both flows. Doing: > > > > > > > > m=audio 20000 RTP/AVP 0 8 > > > > > > > > on the other hand only implies that you book the ceil() > > > > of both flows. > > > > > > > > Why is it so advantageous to segregate these into two > > > > streams? The implicit coupling you're after seems like > > > > it's not well handled by SDP right now, and I'd bet > > > > that this would be better taken up as a working item > > > > for SDPng (if not already there) because my understanding > > > > is that explicit audio synch with video is also problematic in > > > > SDP right now, which is sort of similar. > > > > > > > > Mike > > > > > > > > -- > > Gonzalo Camarillo Phone : +358 9 299 33 71 > > Oy L M Ericsson Ab Mobile: +358 40 702 35 35 > > Telecom R&D Fax : +358 9 299 30 52 > > FIN-02420 Jorvas Email : Gonzalo.Camarillo@ericsson.com > > Finland http://www.hut.fi/~gonzalo > > > > --- > Jonathan D. Rosenberg 72 Eagle Rock Ave. > Chief Scientist First Floor > dynamicsoft East Hanover, NJ 07936 > jdrosen@dynamicsoft.com FAX: (973) 952-5050 > http://www.cs.columbia.edu/~jdrosen PHONE: (973) 952-5000 > http://www.dynamicsoft.com > > _______________________________________________ > SIP mailing list > SIP@lists.bell-labs.com > http://lists.bell-labs.com/mailman/listinfo/sip -- Gonzalo Camarillo Phone : +358 9 299 33 71 Oy L M Ericsson Ab Mobile: +358 40 702 35 35 Telecom R&D Fax : +358 9 299 31 18 FIN-02420 Jorvas Email : Gonzalo.Camarillo@ericsson.com Finland
- draft-camarillo-sip-sdp-00.txt Culpepper, Bert
- Re: [SIP] draft-camarillo-sip-sdp-00.txt Gonzalo Camarillo
- Re: [SIP] draft-camarillo-sip-sdp-00.txt Gonzalo Camarillo
- Re: [SIP] draft-camarillo-sip-sdp-00.txt Michael Thomas
- RE: [SIP] draft-camarillo-sip-sdp-00.txt Culpepper, Bert
- Re: [SIP] draft-camarillo-sip-sdp-00.txt Gonzalo Camarillo
- RE: [SIP] draft-camarillo-sip-sdp-00.txt Jonathan Rosenberg
- Re: [SIP] draft-camarillo-sip-sdp-00.txt Gonzalo Camarillo