20 Feb
2007
20 Feb
'07
8:23 a.m.
"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.
Regards,
Michel.
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.
regards
klaus
Klaus Darilion wrote:
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).
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? 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. >
> 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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>