[Users] SDP Parser

[Klaus Darilion]

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.

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.

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
>>>>
>>>>
>>
>>


-- 
Klaus Darilion
nic.at