Implementing Paging on the Linksys SPA9XX phones

John Lange wrote:

> I have created this request in the bug-tracker which is
probably a
> better way to comment rather than this list.
> 
> Link here: http://bugs.d
igium.com/view.php?id=7172

No, it most definitely is not. The bug tracker is not a
discussion
forum, and the bug posting guidelines clearly state that.

Adding support for this to Asterisk will be non-trivial, to
say the least.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

On Wed, 2006-05-17 at 10:48 -0500, Kevin P. Fleming wrote:

> Adding support for this to Asterisk will be
non-trivial, to say the least.

Perhaps, and I'm certainly not in the best position to
judge this but
here is my thinking;

>From what I can see in the packet capture, only a single
UDP packet sent
to a multicast address is required to signal the phones that
a page is
coming. Then RTP packets are sent to a multicast address for
the actual
audio of the page.

RTP is already a part of Asterisk so at least it should be
possible to
take this code and implement it in a new paging app which
could also do
the UDP.

In any case, even if this is non-trivial I believe the
effort would be
well worth it to finally have a "real" paging
system in Asterisk. I have
a feeling some other SIP phone makers would also incorporate
this method
into their firmware if it was supported.

-- 
John Lange


_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

John Lange wrote:

> In any case, even if this is non-trivial I believe the
effort would be
> well worth it to finally have a "real"
paging system in Asterisk. I have
> a feeling some other SIP phone makers would also
incorporate this method
> into their firmware if it was supported.

Other SIP phone manufacturers already support multicast
using the
traditional SIP/SDP model (no 'special' multicast packets
like
Cisco/Linksys use).

The bigger issue, though, is that applications in Asterisk
are
completely unaware of the channel types they are running on
(generally).
This means that the Page application can't generate
'multicast RTP',
since it has no idea whether RTP is even in use or not (and
it may be in
use on some channels in the paging group and not others).
Any solution
will require finding a way to abstract a group of phones
listening
(only) to the same audio stream and learning whether they
support
multicast or not... not trivial, as I said 
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

On Wed, 2006-05-17 at 13:32 -0500, Kevin P. Fleming wrote:
> John Lange wrote:
> 
> > In any case, even if this is non-trivial I believe
the effort would be
> > well worth it to finally have a "real"
paging system in Asterisk. I have
> > a feeling some other SIP phone makers would also
incorporate this method
> > into their firmware if it was supported.
> 
> Other SIP phone manufacturers already support multicast
using the
> traditional SIP/SDP model (no 'special' multicast
packets like
> Cisco/Linksys use).

Can you give examples of which phones? I don't know what
you mean by the
SIP/SDP model and google on those terms didn't turn up
anything
meaningful with regard to paging.

> The bigger issue, though, is that applications in
Asterisk are
> completely unaware of the channel types they are
running on (generally).
> This means that the Page application can't generate
'multicast RTP',
> since it has no idea whether RTP is even in use or not
(and it may be in
> use on some channels in the paging group and not
others). Any solution
> will require finding a way to abstract a group of
phones listening
> (only) to the same audio stream and learning whether
they support
> multicast or not... not trivial, as I said 

It may be as simple as specifying what type of
"paging" a given device
supports (if any) in the config files for the device. If a
given device
is then included in a page group then asterisk would handle
each device
in its own way.

So for example; SIP phones which only support paging because
one of
their lines is set to auto-answer (like the ciscos) can be
one setting;
phones that support multi-cast linksys style is another;
SIP/SDP yet
another; and so on.

We are very interested in solving this issue because paging
seems to be
the most often asked for feature in Asterisk that doesn't
really exist.

-- 
John Lange


_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

John Lange wrote:

> Can you give examples of which phones? I don't know
what you mean by the
> SIP/SDP model and google on those terms didn't turn up
anything
> meaningful with regard to paging.

Snom, for certain, supports multicast RTP, but it has to be
done using
the normal SIP/SDP method of sending an INVITE to the phone
and telling
it what multicast stream it should listen to (and whether it
should
auto-answer, etc.).

> It may be as simple as specifying what type of
"paging" a given device
> supports (if any) in the config files for the device.
If a given device
> is then included in a page group then asterisk would
handle each device
> in its own way.

No, that is the simple part, because it's in the channel
drivers.
Channel drivers don't know what 'paging' is, they only
understand
channels, which can be attached to anything at all.

> So for example; SIP phones which only support paging
because one of
> their lines is set to auto-answer (like the ciscos) can
be one setting;
> phones that support multi-cast linksys style is
another; SIP/SDP yet
> another; and so on.
> 
> We are very interested in solving this issue because
paging seems to be
> the most often asked for feature in Asterisk that
doesn't really exist.

It's not. The most asked for feature is key-system
functionality,
followed closed by T.38 media gateway support. Doing paging
the way we
do it now seems to work for many people.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Not wishing necessarily to open that Pandora's box but I
can't help 
myself....
> John Lange wrote:
>
>   
>> We are very interested in solving this issue
because paging seems to be
>> the most often asked for feature in Asterisk that
doesn't really exist.
>>     

Kevin P. Fleming replied:
> It's not. The most asked for feature is key-system
functionality,
> followed closed by T.38 media gateway support. Doing
paging the way we
> do it now seems to work for many people.
>   

While not disagreeing with Mr. Fleming's opinion, I would
like to know 
where reliable SIP jitter buffering is on the radar.
Obviously rarely an 
issue for those deploying SIP devices within a LAN but those
of us 
offering hosted Asterisk PBX services really notice the lack
of SIP 
jitter buffering (as would anyone trying to support remote
users on 
their Asterisk boxes).

g.

-- 
George Pajari, netVOICE communications    604 484 VOIP (484
8647 x102)
Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638
8647 x102)
                  www.netvoice.ca  www.ip-centrex.ca
      www.digium.ca www.grandstream.ca www.sipura.ca
www.snom.ca

_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

On Wed, 2006-05-17 at 17:22 -0500, Kevin P. Fleming wrote:
> John Lange wrote:
> 
> > Can you give examples of which phones? I don't
know what you mean by the
> > SIP/SDP model and google on those terms didn't
turn up anything
> > meaningful with regard to paging.
> 
> Snom, for certain, supports multicast RTP, but it has
to be done using
> the normal SIP/SDP method of sending an INVITE to the
phone and telling
> it what multicast stream it should listen to (and
whether it should
> auto-answer, etc.).

This could be what the Linksys phones are doing as well but
I don't know
enough about the protocol to say for certain. In any case
Asterisk
doesn't support it and it would be nice if it did. 

> > We are very interested in solving this issue
because paging seems to be
> > the most often asked for feature in Asterisk that
doesn't really exist.
> 
> It's not. The most asked for feature is key-system
functionality,
> followed closed by T.38 media gateway support.

If by key-system functionality you mean line indicator
lights when
people are on the phone then I second that. It is also very
often
requested.

T.38 media gateway support would be nice but there are so
many other
ways to solve this issue I don't consider it a priority.
I've never had
that request. People just use analog lines for faxing in
larger offices
or use a digital fax gateway.

> Doing paging the way we
> do it now seems to work for many people.

It can work but its not very scalable. Can Asterisk scale to
a 100 phone
conference call in fractions of a second? And on phones that
don't
support the auto-answer sip header its a bit of disaster
requiring more
expensive multi-line phones in situations that otherwise
wouldn't need
them.

Never the less I believe the current implementation will
work well for
the SPA9XX phones in the specific case where we most need it
since there
is only about 30 handsets.
---

Anyhow thats a bit off topic. I'm still hoping to get this
implemented
in Asterisk somehow and if anyone else is interested in
doing some
analysis on whats required to make it happen please let me
know.

-- 
John Lange


_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

18 maj 2006 kl. 03.49 skrev George Pajari:

> Not wishing necessarily to open that Pandora's box but
I can't help  
> myself....
>> John Lange wrote:
>>
>>
>>> We are very interested in solving this issue
because paging seems  
>>> to be
>>> the most often asked for feature in Asterisk
that doesn't really  
>>> exist.
>>>
>
> Kevin P. Fleming replied:
>> It's not. The most asked for feature is key-system
functionality,
>> followed closed by T.38 media gateway support.
Doing paging the  
>> way we
>> do it now seems to work for many people.
>>
>
> While not disagreeing with Mr. Fleming's opinion, I
would like to  
> know where reliable SIP jitter buffering is on the
radar. Obviously  
> rarely an issue for those deploying SIP devices within
a LAN but  
> those of us offering hosted Asterisk PBX services
really notice the  
> lack of SIP jitter buffering (as would anyone trying to
support  
> remote users on their Asterisk boxes).

We have had one in the bug tracker for a long time,
requesting  
testing many, many times and got very little
feedback. I do thank you for dedicating time to test this
this week  
and next, so we can get some valuable
input and possibly move this patch forward.

It's available both in test-this-branch and the
jitterbuffer branch.  
You enable it in the Makefile (we're waiting for Russell
to assist us in making it configurable in the new config
system).

/Olle
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Olle E Johansson wrote:
> It's available both in test-this-branch and the
jitterbuffer branch. You 
> enable it in the Makefile (we're waiting for Russell
> to assist us in making it configurable in the new
config system).

I did that last week.  It is available as an option in
"make menuselect" in the 
rtpjitterbuffer branch.

Russell
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

George Pajari wrote:

> While not disagreeing with Mr. Fleming's opinion, I
would like to know
> where reliable SIP jitter buffering is on the radar.
Obviously rarely an
> issue for those deploying SIP devices within a LAN but
those of us
> offering hosted Asterisk PBX services really notice the
lack of SIP
> jitter buffering (as would anyone trying to support
remote users on
> their Asterisk boxes).

I did not state an opinion, nor did I state what were the
most useful
things to get added to Asterisk. The poster mentioned 'the
most
requested feature' and I responded to that.

Yes, an RTP jitter buffer would be a welcome thing (there is
no jitter
buffer for SIP, it's signaling!). However, until the code
is stable and
we get solid, reliable test reports, it's not going to get
merged, no
matter how badly we want it. An unreliable, buggy or
unmaintainable
jitter buffer is worse than none at all.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev