[Users] SDP Parser

Michel Bensoussan michel at extricom.com
Wed Feb 21 16:12:07 CET 2007


Hi Ovidiu

Thanks for your help.

The SIP proxy seems to me a better approach. I think I'll use also 
dialog module (if I success using it) and later mediaproxy.

Michel.

Ovidiu Sas wrote:
> Hi Michel,
>
> It all depends on what you want to do.  Here are the two approaches
> and the associated issues:
>
> 1. SIP Proxy:
>  - you don't have control over the negotiated codec(s)
>  - if the two parties agrees on a set of codecs, you don't know which
> one will be used.  Even worst, the parties may change the codec
> without renegotiating, since it is part of the negotiated set.
>  - you don't have control over the stream packetization.
>
> 2. B2BUA:
>  - you may force a single code to be used.
>  - you may force the packetization for outgoing media, but you have
> no control over the packetization of the incoming stream.
>
>
> Computing the required bandwidth for VoIP calls is really tough unless
> you work with well known SIP UAs and you know what to expect.
>
>
>
> Hope this helps,
> Ovidiu Sas
>
>
>
> PS: you might wanna take a look at http://www.nslu2-linux.org/
> There are several VoIP packages (openSER, rtpproxy, asterisk) that
> were ported to several embedded platforms including dd-wrt.
>
>
> On 2/20/07, Michel Bensoussan <michel at extricom.com> wrote:
>> Klaus Darilion wrote:
>> > Michel Bensoussan wrote:
>> >> "If the media goes directly from caller to callee I wonder why you
>> >> need to know the bandwidth at all as the RTP packets may be out of
>> >> your network."
>> >>
>> >> I need to write a CAC (Call Admission Control) module for an 802.11
>> >> AP (Access Point).
>> >> The idea is to use a SIP Proxy to monitoring bandwith utilization
>> >> according to codec, and allow or disallow new sessions, depending on
>> >> resources.
>> >
>> > Do you need to know the bandwidth in advance (thus guessing the neede
>> > bandwidth during call setup and eventually deny the call setup if the
>> > required bandwidth can not be guaranteed) (a) or do you need to know
>> > the exact current bandwidth need (b)?
>> >
>> > In case of (a) I think you need a B2BUA in the AP.
>> I just look for B2BUA in google (I didn't know it) and it's interesting.
>> Guessing the bandwidth can be good for a first step but this is not 
>> enough.
>>
>> >
>> > In case of (b) you can parse the RTP sockets and then count the media
>> > packets routed by the AP to calculate the bandwidth.
>> >
>> > Or you can port mediaproxy to the accesspoint and use thus 
>> capabilities.
>> Case (b) is of course the target and mediaproxy looks useful, even if I
>> don't need the NAT features.
>> >
>> > regards
>> > klaus
>> >
>> >
>> >>
>> >> Regards,
>> >> Michel.
>> >>
>> >> Klaus Darilion wrote:
>> >>> Michel Bensoussan wrote:
>> >>>> Klaus Darilion wrote:
>> >>>>> Parsing the SDP does not give you the used codec as there may be
>> >>>>> several codecs in the SDP and you do not know which codec is used
>> >>>>> by the clients.
>> >>>> This is true for INVITE message but as I understand (but I'm not
>> >>>> familiar with SIP), in the OK message, we can determine which codec
>> >>>> is used. No?
>> >>>
>> >>> Not always. Often the 200 OK contains only one codec which will be
>> >>> used by both parties. But I think there may also be asynchronous
>> >>> codec (caller sends G711, callee sends G729).
>> >>>
>> >>>>> But for example you can use mediaproxy. Mediaproxy allows you to
>> >>>>> retrieve the status of all current calls (codecs, bandwidth, ...)
>> >>>> Well, the mediaproxy module needs an external proxy server. So it
>> >>>> seems to be too heavy for my needs.
>> >>>> The real time session statistics (from MediaProxy Server) will be
>> >>>> very useful but I'm not sure it's a good idea to use the server it
>> >>>> if I don't need the NAT traversal features.
>> >>>
>> >>> If the media goes directly from caller to callee I wonder why you
>> >>> need to know the bandwidth at all as the RTP packets may be out of
>> >>> your network.
>> >>>
>> >>> regards
>> >>> klaus
>> >>>
>> >>>>>
>> >>>>> regards
>> >>>>> klaus
>> >>>>>
>> >>>>> Michel Bensoussan wrote:
>> >>>>>> Hello
>> >>>>>> For each voice session I need to know the used codec (for
>> >>>>>> bandwith calculation). For that I need to parse the SIP message
>> >>>>>> body.
>> >>>>>> I didn't find in OpenSER such a functionality.
>> >>>>>> Is there a module that doing that?
>> >>>>>> Or maybe someone is working on it?
>> >>>>>> A suggestion for an open source?
>> >>>>>>
>> >>>>>> Thanks.
>> >>>>>>
>> >>>>>> Regards,
>> >>>>>> Michel.
>> >>>>>>
>> >>>>>>
>> >>>>>>
>> >>>>>> _______________________________________________
>> >>>>>> Users mailing list
>> >>>>>> Users at openser.org
>> >>>>>> http://openser.org/cgi-bin/mailman/listinfo/users
>> >>>>>
>> >>>>>
>> >>>
>> >>>
>> >
>> >
>>
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