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Douglas Garstang

2006-Mar-07 09:12 UTC

[Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

I have a configuration where RTP traffic is going out interface pub0, and coming
back into through pub1.
I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an shows:
 
udp        0    788 0.0.0.0:5060            0.0.0.0:*
 
which means that Asterisk is listening on all addresses (on all interfaces?).
 
Anyway, when the RTP traffic comes back in on interface pub0, Asterisk does
nothing with it. A 'rtp debug' shows it's receiving the RTP packets,
it just seems it does nothing with them.
 
Anyone seen this?
 
Doug.
 
 
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Robert Webb

2006-Mar-07 09:21 UTC

head link

[Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

On Tue, 7 Mar 2006 09:12:25 -0700
  "Douglas Garstang" <dgarstang@oneeighty.com>
wrote:> I have a configuration where RTP traffic is going out 
>interface pub0, and coming back into through pub1.
> I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an 
>shows:
> 
> udp        0    788 0.0.0.0:5060            0.0.0.0:*
> 
> which means that Asterisk is listening on all addresses 
>(on all interfaces?).
> 
> Anyway, when the RTP traffic comes back in on interface 
>pub0, Asterisk does nothing with it. A 'rtp debug' shows 
>it's receiving the RTP packets, it just seems it does 
>nothing with them.
> 
> Anyone seen this?
> 
> Doug.
> 
> 
I thought all RTP was controlled through rtp.conf and only 
the SIP traffic was controlled through SIP.conf. I am not 
sure what settings, beside the RTP port range, you can out 
into the rtp.conf though.

Robert

Alexander Lopez

2006-Mar-07 09:38 UTC

head link

[Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

Asterisk does not like multiple interfaces in the way you are configured. You
can either:

A) use the bindaddr in the sip.conf to limit where the packsge come and go.

B) use an outside traffic manager

Look up the archives, kpf explained why this would not work, as asterisk
can't do load balancing at this time


-----Original Message-----
From: "Robert Webb" <asterisk@ropeguru.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Sent: 3/7/06 11:27 AM
Subject: Re: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic


On Tue, 7 Mar 2006 09:12:25 -0700
  "Douglas Garstang" <dgarstang@oneeighty.com>
wrote:> I have a configuration where RTP traffic is going out 
>interface pub0, and coming back into through pub1.
> I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an 
>shows:
> 
> udp        0    788 0.0.0.0:5060            0.0.0.0:*
> 
> which means that Asterisk is listening on all addresses 
>(on all interfaces?).
> 
> Anyway, when the RTP traffic comes back in on interface 
>pub0, Asterisk does nothing with it. A 'rtp debug' shows 
>it's receiving the RTP packets, it just seems it does 
>nothing with them.
> 
> Anyone seen this?
> 
> Doug.
> 
> 
I thought all RTP was controlled through rtp.conf and only 
the SIP traffic was controlled through SIP.conf. I am not 
sure what settings, beside the RTP port range, you can out 
into the rtp.conf though.

Robert
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Douglas Garstang

2006-Mar-07 09:45 UTC

head link

[Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

Pardon my candour, but for a product Digium calls 'enterprise grade' it
sure seems to be missing a few features.

-----Original Message-----
From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com]
Sent: Tuesday, March 07, 2006 9:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
traffic


Asterisk does not like multiple interfaces in the way you are configured. You
can either:

A) use the bindaddr in the sip.conf to limit where the packsge come and go.

B) use an outside traffic manager

Look up the archives, kpf explained why this would not work, as asterisk
can't do load balancing at this time


-----Original Message-----
From: "Robert Webb" <asterisk@ropeguru.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Sent: 3/7/06 11:27 AM
Subject: Re: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic


On Tue, 7 Mar 2006 09:12:25 -0700
  "Douglas Garstang" <dgarstang@oneeighty.com>
wrote:> I have a configuration where RTP traffic is going out 
>interface pub0, and coming back into through pub1.
> I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an 
>shows:
> 
> udp        0    788 0.0.0.0:5060            0.0.0.0:*
> 
> which means that Asterisk is listening on all addresses 
>(on all interfaces?).
> 
> Anyway, when the RTP traffic comes back in on interface 
>pub0, Asterisk does nothing with it. A 'rtp debug' shows 
>it's receiving the RTP packets, it just seems it does 
>nothing with them.
> 
> Anyone seen this?
> 
> Doug.
> 
> 
I thought all RTP was controlled through rtp.conf and only 
the SIP traffic was controlled through SIP.conf. I am not 
sure what settings, beside the RTP port range, you can out 
into the rtp.conf though.

Robert
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Alexander Lopez

2006-Mar-07 15:34 UTC

head link

[Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

To retort, Digium has ever to my knowledge, stamped an 'Enterprise
Grade' mark on the product.  If you are worried about a single point of
failure you may want to replace your toaster.

Asterisk is missing a 'few features' no doubt about it, but it is open
source, it will be a welcome addition if you would like to add
multi-homing support in, might as well do media multi-homing with call
diversity. This will definably be a non-trivial re-architecture of the
core.

The 'missing a few features' way of thinking is what has made Asterisk
what it is today.
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Douglas Garstang
> Sent: Tuesday, March 07, 2006 11:46 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
traffic> 
> Pardon my candour, but for a product Digium calls 'enterprise
grade'
it> sure seems to be missing a few features.
> 
> -----Original Message-----
> From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com]
> Sent: Tuesday, March 07, 2006 9:39 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
> traffic
> 
> 
> Asterisk does not like multiple interfaces in the way you are
configured.> You can either:
> 
> A) use the bindaddr in the sip.conf to limit where the packsge come
and> go.
> 
> B) use an outside traffic manager
> 
> Look up the archives, kpf explained why this would not work, as
asterisk> can't do load balancing at this time
> 
> 
> -----Original Message-----
> From: "Robert Webb" <asterisk@ropeguru.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-> users@lists.digium.com>
> Sent: 3/7/06 11:27 AM
> Subject: Re: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
traffic> 
> 
> On Tue, 7 Mar 2006 09:12:25 -0700
>   "Douglas Garstang" <dgarstang@oneeighty.com> wrote:
> > I have a configuration where RTP traffic is going out
> >interface pub0, and coming back into through pub1.
> > I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an
> >shows:
> >
> > udp        0    788 0.0.0.0:5060            0.0.0.0:*
> >
> > which means that Asterisk is listening on all addresses
> >(on all interfaces?).
> >
> > Anyway, when the RTP traffic comes back in on interface
> >pub0, Asterisk does nothing with it. A 'rtp debug' shows
> >it's receiving the RTP packets, it just seems it does
> >nothing with them.
> >
> > Anyone seen this?
> >
> > Doug.
> >
> >
> 
> I thought all RTP was controlled through rtp.conf and only
> the SIP traffic was controlled through SIP.conf. I am not
> sure what settings, beside the RTP port range, you can out
> into the rtp.conf though.
> 
> Robert
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

Douglas Garstang

2006-Mar-08 08:18 UTC

head link

[Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

I can't be bothered looking for the link right now, but it's definitely
stated somewhere on Digium's website.

-----Original Message-----
From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com]
Sent: Tuesday, March 07, 2006 3:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
traffic


To retort, Digium has ever to my knowledge, stamped an 'Enterprise
Grade' mark on the product.  If you are worried about a single point of
failure you may want to replace your toaster.

Asterisk is missing a 'few features' no doubt about it, but it is open
source, it will be a welcome addition if you would like to add
multi-homing support in, might as well do media multi-homing with call
diversity. This will definably be a non-trivial re-architecture of the
core.

The 'missing a few features' way of thinking is what has made Asterisk
what it is today.
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Douglas Garstang
> Sent: Tuesday, March 07, 2006 11:46 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
traffic> 
> Pardon my candour, but for a product Digium calls 'enterprise
grade'
it> sure seems to be missing a few features.
> 
> -----Original Message-----
> From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com]
> Sent: Tuesday, March 07, 2006 9:39 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
> traffic
> 
> 
> Asterisk does not like multiple interfaces in the way you are
configured.> You can either:
> 
> A) use the bindaddr in the sip.conf to limit where the packsge come
and> go.
> 
> B) use an outside traffic manager
> 
> Look up the archives, kpf explained why this would not work, as
asterisk> can't do load balancing at this time
> 
> 
> -----Original Message-----
> From: "Robert Webb" <asterisk@ropeguru.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-> users@lists.digium.com>
> Sent: 3/7/06 11:27 AM
> Subject: Re: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
traffic> 
> 
> On Tue, 7 Mar 2006 09:12:25 -0700
>   "Douglas Garstang" <dgarstang@oneeighty.com> wrote:
> > I have a configuration where RTP traffic is going out
> >interface pub0, and coming back into through pub1.
> > I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an
> >shows:
> >
> > udp        0    788 0.0.0.0:5060            0.0.0.0:*
> >
> > which means that Asterisk is listening on all addresses
> >(on all interfaces?).
> >
> > Anyway, when the RTP traffic comes back in on interface
> >pub0, Asterisk does nothing with it. A 'rtp debug' shows
> >it's receiving the RTP packets, it just seems it does
> >nothing with them.
> >
> > Anyone seen this?
> >
> > Doug.
> >
> >
> 
> I thought all RTP was controlled through rtp.conf and only
> the SIP traffic was controlled through SIP.conf. I am not
> sure what settings, beside the RTP port range, you can out
> into the rtp.conf though.
> 
> Robert
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Douglas Garstang

2006-Mar-08 16:12 UTC

head link

[Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

Asterisk calls the Business Edition 'enterprise grade'. It's right
there on the Digium website. It's the same dang code as the open source
version, just older.

-----Original Message-----
From: Matt Riddell [NZ] [mailto:matt.riddell@sineapps.com]
Sent: Tuesday, March 07, 2006 6:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
traffic

Douglas Garstang wrote:> Pardon my candour, but for a product Digium calls 'enterprise
grade' it sure seems to be missing a few features.
Um...it's Open Source.  Why don't you add the features you require
yourself or pay someone to add them for you...

This is your third similar post in as many days.

-- 
Cheers,

Matt Riddell
_______________________________________________

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
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Michael Collins

2006-Mar-08 18:06 UTC

head link

[Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

For the record, Douglas is correct on this point of "enterprise-grade"
being on ABE:
http://www.digium.com/index.php?menu=product_category&category=software

Copied and pasted right from the website, it says:

Asterisk Business Edition(tm)
Digium(tm), the leader in open source telephony, offers Asterisk
Business Edition, an enterprise-grade version of its acclaimed open
source PBX for the Linux operating system. This version provides tested
reliability of critical functions and features, tailored for small- and
medium-sized business applications.

Now, as to the debate about what is and is not available in an
"enterprise-grade" product, I will have to defer to those who actually
use Asterisk in the enterprise - I only use it for tinkering and minor
voice broadcasting campaigns.

-MC
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Douglas Garstang
> Sent: Wednesday, March 08, 2006 7:19 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
traffic> 
> I can't be bothered looking for the link right now, but it's
definitely> stated somewhere on Digium's website.
> 
> -----Original Message-----
> From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com]
> Sent: Tuesday, March 07, 2006 3:34 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
> traffic
> 
> 
> To retort, Digium has ever to my knowledge, stamped an 'Enterprise
> Grade' mark on the product.  If you are worried about a single point
of> failure you may want to replace your toaster.
> 
> Asterisk is missing a 'few features' no doubt about it, but it is
open
> source, it will be a welcome addition if you would like to add
> multi-homing support in, might as well do media multi-homing with call
> diversity. This will definably be a non-trivial re-architecture of the
> core.
> 
> The 'missing a few features' way of thinking is what has made
Asterisk
> what it is today.
> 
> > -----Original Message-----
> > From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-> > bounces@lists.digium.com] On Behalf Of Douglas Garstang
> > Sent: Tuesday, March 07, 2006 11:46 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
> traffic
> >
> > Pardon my candour, but for a product Digium calls 'enterprise
grade'
> it
> > sure seems to be missing a few features.
> >
> > -----Original Message-----
> > From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com]
> > Sent: Tuesday, March 07, 2006 9:39 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
> > traffic
> >
> >
> > Asterisk does not like multiple interfaces in the way you are
> configured.
> > You can either:
> >
> > A) use the bindaddr in the sip.conf to limit where the packsge come
> and
> > go.
> >
> > B) use an outside traffic manager
> >
> > Look up the archives, kpf explained why this would not work, as
> asterisk
> > can't do load balancing at this time
> >
> >
> > -----Original Message-----
> > From: "Robert Webb" <asterisk@ropeguru.com>
> > To: "Asterisk Users Mailing List - Non-Commercial
Discussion"
> <asterisk-
> > users@lists.digium.com>
> > Sent: 3/7/06 11:27 AM
> > Subject: Re: [Asterisk-Users] Oh this is bad.... bindaddr and rtp
> traffic
> >
> >
> > On Tue, 7 Mar 2006 09:12:25 -0700
> >   "Douglas Garstang" <dgarstang@oneeighty.com> wrote:
> > > I have a configuration where RTP traffic is going out
> > >interface pub0, and coming back into through pub1.
> > > I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an
> > >shows:
> > >
> > > udp        0    788 0.0.0.0:5060            0.0.0.0:*
> > >
> > > which means that Asterisk is listening on all addresses
> > >(on all interfaces?).
> > >
> > > Anyway, when the RTP traffic comes back in on interface
> > >pub0, Asterisk does nothing with it. A 'rtp debug' shows
> > >it's receiving the RTP packets, it just seems it does
> > >nothing with them.
> > >
> > > Anyone seen this?
> > >
> > > Doug.
> > >
> > >
> >
> > I thought all RTP was controlled through rtp.conf and only
> > the SIP traffic was controlled through SIP.conf. I am not
> > sure what settings, beside the RTP port range, you can out
> > into the rtp.conf though.
> >
> > Robert
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

Douglas Garstang

2006-Mar-09 00:18 UTC

head link

[Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

By 'code for asterisk' are you referring to the Asterisk source code? If
so, step back and think about your statement for a moment. If, for Asterisk to
be enterprise class, it's source code needs to be modified from it's
current content, it's hardly enterprise class, is it?
 
If 'code for asterisk' refers to extensions.conf and the like, I fail to
see how anything within the asterisk dial plan would account for the apparent
inability of asterisk to listen for RTP traffic on all network interfaces.
 
Doug.

	-----Original Message----- 
	From: Matt Riddell [NZ] [mailto:matt.riddell@sineapps.com] 
	Sent: Wed 3/8/2006 11:43 PM 
	To: Asterisk Users Mailing List - Non-Commercial Discussion 
	Cc: 
	Subject: Re: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic
	
	

	Douglas Garstang wrote:
	> Asterisk calls the Business Edition 'enterprise grade'. It's
right there on the Digium website. It's the same dang code as the open
source version, just older.
	
	We are using it successfully in quite a few enterprise roll outs.  If
	you are unable to, maybe you should attend one of our training sessions,
	which among other things discuss how to code for Asterisk.
	
	If however you'd rather just complain, please do so to /dev/null
	
	--
	Cheers,
	
	Matt Riddell
	_______________________________________________
	
	http://www.sineapps.com/news.php (Daily Asterisk News - html)
	http://freevoip.gedameurope.com (Free Asterisk Voip Community)
	http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
	_______________________________________________
	--Bandwidth and Colocation provided by Easynews.com --
	
	Asterisk-Users mailing list
	To UNSUBSCRIBE or update options visit:
	   http://lists.digium.com/mailman/listinfo/asterisk-users

Koopmann, Jan-Peter

2006-Mar-09 00:46 UTC

head link

[Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

On Thursday, March 09, 2006 8:18 AM Douglas Garstang wrote: 
> By 'code for asterisk' are you referring to the Asterisk source
code?
> If so, step back and think about your statement for a moment. If, for
> Asterisk to be enterprise class, it's source code needs to be
> modified from it's current content, it's hardly enterprise class,
is
> it?    
What is this thread all about? Is Asterisk "enterprise class"? The
answer is
obvious: It depends on your definition of "enterprise class".

If you definition includes things like "RTP in/out traffic on multiple
interfaces must work" then the answer is no.

If you definition is somewhere along the line "can be used in most
enterprises without problems" then the answer is yes.

If you need a feature you at least have the possibility to code it yourself
(yes, source code). Avaya&Co give you the opportunity to hand in a feature
request but nothing more. Unless you pay for the feature they will probably
not implement it and you have no way of doing so yourself. 

Asterisk does not really meet my personal definition of "enterprise
class"
but since there is no commonly accepted definition in the first place, why
trust Digiums words on the website at all? I usually do not trust any
marketing phrase like that no matter what the product is. Try Asterisk
yourself and if your decision is that you cannot use it, then don't! But
please stop getting on peoples nerves bashing on the term "enterprise
class". It will not get you or us anywhere.

If you are not satisfied with what Asterisk can achieve you have plenty of
other choices. Feel free to use them. Feel free to contribute to the
project. Constructive criticism is wanted (at least in most of the cases
this seems to be true even though there is room for improvement, agreed).
Currently you are not helping at all. Annoying is a term that comes to mind
though...

Kind regards,
  JP
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