[Users] SDP Parser

[Michel Bensoussan]

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.
As far as I understand B2BUA is one to one. I need to handle several 
call at a time.
>
> 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.
>>>>>>
>>>>>>
>>>>>>
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>>>>>
>>>>>
>>>
>>>
>
>