22 Aug
2006
22 Aug
'06
8:43 a.m.
Hi,
Is there a way in Ser to stablish a preferred order of codecs to be used
by the phones?
Thanks,
Ricardo.
22 Aug
22 Aug
8:48 a.m.
Hi,
No i dont think so, because SER has nothing to do with audio/video
streaming whatsoever. Besides, this functionality should be attributed
rather to clients than servers.
Andrey.
On 8/22/06, Ricardo Carvalho <rcarvalho(a)iric.up.pt> wrote:
Hi,
Is there a way in Ser to stablish a preferred order of codecs to be used
by the phones?
Thanks,
Ricardo.
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9:02 a.m.
For what I've searched about, I also think Ser isn't capable of that,
however, since I pretend to use Ser as a SIP router in the campus where
I'm deploying this VoIP infrastructure, that functionality should be
attributed to SIP server, so I can force terminals to use less bandwidth
consuming codecs, because bandwidth may be some big issue when we talk
about tens and hundreds of calls simultaneously!
If is there any way to do it with Ser, please let me know.
Thanks,
Ricardo.
Andrey Kouprianov wrote:
Hi,
No i dont think so, because SER has nothing to do with audio/video
streaming whatsoever. Besides, this functionality should be attributed
rather to clients than servers.
Andrey.
On 8/22/06, Ricardo Carvalho <rcarvalho(a)iric.up.pt> wrote:
Hi,
Is there a way in Ser to stablish a preferred order of codecs to be used
by the phones?
Thanks,
Ricardo.
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
_______________________________________________
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9:13 a.m.
On Tue, Aug 22, 2006 at 07:48:34PM +0700, Andrey Kouprianov wrote:
Hi,
No i dont think so, because SER has nothing to do with audio/video
streaming whatsoever. Besides, this functionality should be attributed
rather to clients than servers.
Andrey.
it's true, but we can make something like this:
for example: prefered codec should be "8"
subst(/^m=audio(.*)RTP\/AVP(.*) 8/m=audio\1RTP\/AVP 8\2/i);
On 8/22/06, Ricardo Carvalho <rcarvalho(a)iric.up.pt> wrote:
--
Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info
AD1-UANIC * ICQ: 122351182 * http://www.start4.info
Hi,
Is there a way in Ser to stablish a preferred order of codecs to be used
by the phones?
Thanks,
Ricardo.
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
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9:20 a.m.
What if SIP client on the other end does not support that codec? Of
course, I understand that u and a-law are mandatory for every SIP
client, but let's pretend that some clients just support 1 or maximum
2 codes. This substituting stuff is like playing with fire :P
Andrey.
On 8/22/06, Alexandr Dubovikov <shurik(a)start4.info> wrote:
On Tue, Aug 22, 2006 at 07:48:34PM +0700, Andrey
Kouprianov wrote:
Hi,
No i dont think so, because SER has nothing to do with audio/video
streaming whatsoever. Besides, this functionality should be attributed
rather to clients than servers.
Andrey.
it's true, but we can make something like this:
for example: prefered codec should be "8"
subst(/^m=audio(.*)RTP\/AVP(.*) 8/m=audio\1RTP\/AVP 8\2/i);
On 8/22/06, Ricardo Carvalho <rcarvalho(a)iric.up.pt> wrote:
--
Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info
AD1-UANIC * ICQ: 122351182 * http://www.start4.info
Hi,
Is there a way in Ser to stablish a preferred order of codecs to be used
by the phones?
Thanks,
Ricardo.
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers 
9:42 a.m.
Andrey Kouprianov wrote:
What if SIP client on the other end does not support
that codec? Of
course, I understand that u and a-law are mandatory for every SIP
they are not mandatory. they are usually supported, but not mandatory.
Think about an IMS phone, which uses SIP and supports only AMR and GSM
codec.
regards
klaus
client, but let's pretend that some clients just
support 1 or maximum
2 codes. This substituting stuff is like playing with fire :P
Andrey.
On 8/22/06, Alexandr Dubovikov <shurik(a)start4.info> wrote:
On Tue, Aug 22, 2006 at 07:48:34PM +0700, Andrey
Kouprianov wrote:
--
Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info
AD1-UANIC * ICQ: 122351182 * http://www.start4.info
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers Hi,
No i dont think so, because SER has nothing to do with audio/video
streaming whatsoever. Besides, this functionality should be attributed
rather to clients than servers.
Andrey.
it's true, but we can make something like this:
for example: prefered codec should be "8"
subst(/^m=audio(.*)RTP\/AVP(.*) 8/m=audio\1RTP\/AVP 8\2/i);
On 8/22/06, Ricardo Carvalho <rcarvalho(a)iric.up.pt> wrote:
>Hi,
>
>Is there a way in Ser to stablish a preferred order of codecs to be
used
by the
phones?
Thanks,
Ricardo.
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
9:46 a.m.
On Tue, Aug 22, 2006 at 08:20:10PM +0700, Andrey Kouprianov wrote:
What if SIP client on the other end does not support
that codec? Of
course, I understand that u and a-law are mandatory for every SIP
client, but let's pretend that some clients just support 1 or maximum
2 codes. This substituting stuff is like playing with fire :P
in this case subst will not works, because we are looking at " 8" payload.
anyway he can at first make a search and after it make a replace (subst) :-)
Andrey.
>
>subst(/^m=audio(.*)RTP\/AVP(.*) 8/m=audio\1RTP\/AVP 8\2/i);
^^^^
Wbr,
--
Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info
AD1-UANIC * ICQ: 122351182 * http://www.start4.info
9:56 a.m.
i agree with the "search first" part :)
On 8/22/06, Alexandr Dubovikov <shurik(a)start4.info> wrote:
On Tue, Aug 22, 2006 at 08:20:10PM +0700, Andrey
Kouprianov wrote:
What if SIP client on the other end does not
support that codec? Of
course, I understand that u and a-law are mandatory for every SIP
client, but let's pretend that some clients just support 1 or maximum
2 codes. This substituting stuff is like playing with fire :P
in this case subst will not works, because we are looking at " 8" payload.
anyway he can at first make a search and after it make a replace (subst) :-)
Andrey.
>
>subst(/^m=audio(.*)RTP\/AVP(.*) 8/m=audio\1RTP\/AVP 8\2/i);
^^^^
Wbr,
--
Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info
AD1-UANIC * ICQ: 122351182 * http://www.start4.info
11:24 a.m.
Hi guys,
I dont think subst would do anyways. because you need to recalculate the
content-length.
Ser doesBnt (as far as I know) recount content length automaticly. you
need to use some function wich is not exported to the config.
-Atle
* Andrey Kouprianov <andrey.kouprianov(a)gmail.com> [060822 15:56]:
i agree with the "search first" part :)
On 8/22/06, Alexandr Dubovikov <shurik(a)start4.info> wrote:
On Tue, Aug 22, 2006 at 08:20:10PM +0700, Andrey
Kouprianov wrote:
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
What if SIP client on the other end does not
support that codec? Of
course, I understand that u and a-law are mandatory for every SIP
client, but let's pretend that some clients just support 1 or maximum
2 codes. This substituting stuff is like playing with fire :P
in this case subst will not works, because we are looking at " 8" payload.
anyway he can at first make a search and after it make a replace (subst) :-)
Andrey.
>
>subst(/^m=audio(.*)RTP\/AVP(.*) 8/m=audio\1RTP\/AVP 8\2/i);
^^^^
Wbr,
--
Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info
AD1-UANIC * ICQ: 122351182 * http://www.start4.info
11:35 a.m.
As long as you are NOT adding any new entries into SDP content, but
just arranging them in preferred order, I dont think recalculating
content length is strictly necessary.
On 8/22/06, Atle Samuelsen <clona(a)cyberhouse.no> wrote:
Hi guys,
I dont think subst would do anyways. because you need to recalculate the
content-length.
Ser doesBnt (as far as I know) recount content length automaticly. you
need to use some function wich is not exported to the config.
-Atle
* Andrey Kouprianov <andrey.kouprianov(a)gmail.com> [060822 15:56]:
i agree with the "search first" part
:)
On 8/22/06, Alexandr Dubovikov <shurik(a)start4.info> wrote:
On Tue, Aug 22, 2006 at 08:20:10PM +0700, Andrey
Kouprianov wrote:
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
What if SIP client on the other end does not
support that codec? Of
course, I understand that u and a-law are mandatory for every SIP
client, but let's pretend that some clients just support 1 or maximum
2 codes. This substituting stuff is like playing with fire :P
in this case subst will not works, because we are looking at " 8" payload.
anyway he can at first make a search and after it make a replace (subst) :-)
Andrey.
>
>subst(/^m=audio(.*)RTP\/AVP(.*) 8/m=audio\1RTP\/AVP 8\2/i);
^^^^
Wbr,
--
Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info
AD1-UANIC * ICQ: 122351182 * http://www.start4.info
1:10 p.m.
On Tue, Aug 22, 2006 at 10:35:53PM +0700, Andrey Kouprianov wrote:
As long as you are NOT adding any new entries into SDP
content, but
just arranging them in preferred order, I dont think recalculating
content length is strictly necessary.
On 8/22/06, Atle Samuelsen <clona(a)cyberhouse.no>
wrote:
>Hi guys,
>
>I dont think subst would do anyways. because you need to recalculate the
>content-length.
>
>Ser doesBnt (as far as I know) recount content length automaticly. you
>need to use some function wich is not exported to the config.
>
>-Atle
>
of course you don't need to do it, because you just change a position in the
codec's string and the content-length will be the same.
Wbr,
--
Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info
AD1-UANIC * ICQ: 122351182 * http://www.start4.info
1:13 p.m.
Content-Length is recalculated when you modify the body.
Miklos
Andrey Kouprianov wrote:
As long as you are NOT adding any new entries into SDP
content, but
just arranging them in preferred order, I dont think recalculating
content length is strictly necessary.
On 8/22/06, Atle Samuelsen <clona(a)cyberhouse.no> wrote:
Hi guys,
I dont think subst would do anyways. because you need to recalculate the
content-length.
Ser doesBnt (as far as I know) recount content length automaticly. you
need to use some function wich is not exported to the config.
-Atle
* Andrey Kouprianov <andrey.kouprianov(a)gmail.com> [060822 15:56]:
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers i agree with the "search first" part
:)
On 8/22/06, Alexandr Dubovikov <shurik(a)start4.info> wrote:
>On Tue, Aug 22, 2006 at 08:20:10PM +0700, Andrey Kouprianov wrote:
>> What if SIP client on the other end does not support that codec? Of
>> course, I understand that u and a-law are mandatory for every SIP
>> client, but let's pretend that some clients just support 1 or
maximum
>> 2 codes. This substituting stuff is like
playing with fire :P
>
>in this case subst will not works, because we are looking at " 8"
payload.
>anyway he can at first make a search and
after it make a replace
(subst) :-)
>
> Andrey.
>
> >
> >subst(/^m=audio(.*)RTP\/AVP(.*) 8/m=audio\1RTP\/AVP 8\2/i);
^^^^
Wbr,
--
Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info
AD1-UANIC * ICQ: 122351182 * http://www.start4.info
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6568
days inactive
6568
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11 comments
6 participants
participants (6)
-
Alexandr Dubovikov -
Andrey Kouprianov -
Atle Samuelsen -
Klaus Darilion -
Miklos Tirpak -
Ricardo Carvalho