19 Feb
2007
19 Feb
'07
4:07 p.m.
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.
19 Feb
19 Feb
4:32 p.m.
Hello,
I found the open source SDP parser _FreeSDP_ helpful.
Hope it is good for you too.
Best regards,
Anca Vamanu
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(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
4:33 p.m.
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.
But for example you can use mediaproxy. Mediaproxy allows you to
retrieve the status of all current calls (codecs, bandwidth, ...)
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(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
--
Klaus Darilion
nic.at
5:26 p.m.
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?
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.
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
5:36 p.m.
Hi Michel,
actually you cannot find out what is the used codec based on signalling
- each party sends a list of supported codecs (in INVITE and 200 OK),
letting the other party the actual codec to be used for sending media.
AFAIK, the only way is to look into the RTP stream.
Of course, if there is only one advertised codec, there is no mystery ;)/
regards,
bogdan
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?
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:
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
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(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
20 Feb
20 Feb
12:53 p.m.
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.
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users(a)openser.org
>> http://openser.org/cgi-bin/mailman/listinfo/users
>
>
--
Klaus Darilion
nic.at
3:23 p.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.
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users(a)openser.org
>> http://openser.org/cgi-bin/mailman/listinfo/users
>
>
5:30 p.m.
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(a)openser.org
>>>> http://openser.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>
>
--
Klaus Darilion
nic.at
6:01 p.m.
Klaus Darilion wrote:
Michel Bensoussan wrote:
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.
"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.
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(a)openser.org
>>>> http://openser.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>
>
6:24 p.m.
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(a)extricom.com> wrote:
Klaus Darilion wrote:
Michel Bensoussan wrote:
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.
"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.
Case (b)
is of course the target and mediaproxy looks useful, even if I
don't need the NAT features.
regards
klaus
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
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(a)openser.org
>>>> http://openser.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>
>
9:12 p.m.
New subject: [Users] SER behind PIX
Hi everybody. In the topology:
INTERNET
|
PIX (NAT static translate 200.x.x.2 to 10.1.1.2)
|
SER (10.1.1.2)
I can´t register UA from internet.(unauthorized)
The PIX is configured with the "fixup SIP" and "fixup udp SIP"
commands.
I guess there is a problem when UA generates HA1 with the external IP, then
de PIX translates this to internal IP, and the SER can´t authenticate it by
the erroneous hash.
Anybody know some solution for it?
Thanks!
Diego
9:39 p.m.
New subject: [Users] SER behind PIX
Hi Diego,
When you challenge the SIP UA from openSER, provide the realm and
configure the SIP UA to use that particular realm for authorization.
see: http://openser.org/docs/modules/1.2.x/auth_db.html#AEN170
(do not use an empty string for realm)
Regards,
Ovidiu Sas
On 2/20/07, Diego Valencia <dvalencia(a)ip-tel.com.ar> wrote:
Hi everybody. In the topology:
INTERNET
|
PIX (NAT static translate 200.x.x.2 to 10.1.1.2)
|
SER (10.1.1.2)
I can´t register UA from internet.(unauthorized)
The PIX is configured with the "fixup SIP" and "fixup udp SIP"
commands.
I guess there is a problem when UA generates HA1 with the external IP, then
de PIX translates this to internal IP, and the SER can´t authenticate it by
the erroneous hash.
Anybody know some solution for it?
Thanks!
Diego
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
21 Feb
21 Feb
12:20 a.m.
New subject: [Users] SER behind PIX
Hi Ovidiu, I´m currently configured on this way on my openser:
if (!www_authorize("ser.ip-tel.com.ar",
"subscriber")) {
www_challenge("ser.ip-tel.com.ar",
"0");
break;
};
Is it right?
Thanks
Diego
----- Original Message -----
From: "Ovidiu Sas" <sip.nslu(a)gmail.com>
To: "Diego Valencia" <dvalencia(a)ip-tel.com.ar>
Cc: <users(a)openser.org>
Sent: Tuesday, February 20, 2007 4:39 PM
Subject: Re: [Users] SER behind PIX
Hi Diego,
When you challenge the SIP UA from openSER, provide the realm and
configure the SIP UA to use that particular realm for authorization.
see: http://openser.org/docs/modules/1.2.x/auth_db.html#AEN170
(do not use an empty string for realm)
Regards,
Ovidiu Sas
On 2/20/07, Diego Valencia <dvalencia(a)ip-tel.com.ar> wrote:
Hi everybody. In the topology:
INTERNET
|
PIX (NAT static translate 200.x.x.2 to 10.1.1.2)
|
SER (10.1.1.2)
I can´t register UA from internet.(unauthorized)
The PIX is configured with the "fixup SIP" and "fixup udp SIP"
commands.
I guess there is a problem when UA generates HA1 with the external IP,
then
de PIX translates this to internal IP, and the SER can´t authenticate it
by
the erroneous hash.
Anybody know some solution for it?
Thanks!
Diego
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
12:34 a.m.
New subject: [Users] SER behind PIX
yup. Make sure that your subscriber's domain matches the one that you
created in openser db (see subscriber table).
Regards,
Ovidiu Sas
On 2/20/07, Diego Valencia <dvalencia(a)ip-tel.com.ar> wrote:
Hi Ovidiu, I´m currently configured on this way on my
openser:
if (!www_authorize("ser.ip-tel.com.ar",
"subscriber")) {
www_challenge("ser.ip-tel.com.ar",
"0");
break;
};
Is it right?
Thanks
Diego
----- Original Message -----
From: "Ovidiu Sas" <sip.nslu(a)gmail.com>
To: "Diego Valencia" <dvalencia(a)ip-tel.com.ar>
Cc: <users(a)openser.org>
Sent: Tuesday, February 20, 2007 4:39 PM
Subject: Re: [Users] SER behind PIX
Hi Diego,
When you challenge the SIP UA from openSER, provide the realm and
configure the SIP UA to use that particular realm for authorization.
see: http://openser.org/docs/modules/1.2.x/auth_db.html#AEN170
(do not use an empty string for realm)
Regards,
Ovidiu Sas
On 2/20/07, Diego Valencia <dvalencia(a)ip-tel.com.ar> wrote:
Hi everybody. In the topology:
INTERNET
|
PIX (NAT static translate 200.x.x.2 to 10.1.1.2)
|
SER (10.1.1.2)
I can´t register UA from internet.(unauthorized)
The PIX is configured with the "fixup SIP" and "fixup udp SIP"
commands.
I guess there is a problem when UA generates HA1 with the external IP,
then
de PIX translates this to internal IP, and the SER can´t authenticate it
by
the erroneous hash.
Anybody know some solution for it?
Thanks!
Diego
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
5:12 p.m.
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(a)extricom.com> wrote:
Klaus Darilion wrote:
Michel Bensoussan wrote:
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.
"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.
Case (b) is of course the target and mediaproxy looks useful, even if I
don't need the NAT features.
regards
klaus
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
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(a)openser.org
>>>> http://openser.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>
>
20 Feb
20 Feb
6:20 p.m.
Klaus Darilion wrote:
Michel Bensoussan wrote:
As far as I understand B2BUA is
one to one. I need to handle several
call at a time.
"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(a)openser.org
>>>> http://openser.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>
>
6487
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15 comments
6 participants
participants (6)
-
Anca-Maria Vamanu -
Bogdan-Andrei Iancu -
Diego Valencia -
Klaus Darilion -
Michel Bensoussan -
Ovidiu Sas