Siemens Gigaset.Net connection
Nick
Gigaset.Net which allows registered phones to initiate VOIP calls to
each other.
I'm having a few problems getting this to work. It seems that other
users can dial my Gigaset.Net number and connect to me but I can't
connect to them, in fact they can't connect to themselves.
I have checked all phones are registered and they are. So I'm now
leaning towards the view that the service is pants, does anyone else
have any experience.
The G.722 codec that can be used with this service does appear to sound
better than a normal voip call.
www.GymRatZ.co.uk
has one.
You can try calling our gigaset in the shop. You'll get my dull
westcountry tones from the gigaset mailbox but the mailbox is set to
"verh high quality"
Look up "GymRatZ" I think that's the current one. There's an entry for
"GymRatZ UK" but that was set up on the original box that went back due
to faults caused by crap firmware rather than the box.
Try it and see what happens. You won't disturb anyone, the shutters are
down and the alarm is on.
:�)
Tim
> I have checked all phones are registered and they are. So I'm now
> leaning towards the view that the service is pants, does anyone else
> have any experience.
It is a bit pants. Works great if you are a public IP address.
No so good through NAT - has a tendency for 1 way audio.
I'm not fussed by this because I have plenty of other SIP servers to use
which work a lot better.
> The G.722 codec that can be used with this service does appear to sound
> better than a normal voip call.
Indeed. But you can use G.722 with almost any SIP service which doesn't
use asterisk or otherwise limit codecs. Asterisk can support G.722 but
needs patching and a few things aren't quite right. I'm told this will
be addressed in future versions.
I have an SER SIP server I use for internal calls. G.722 is nice but it
is disconcerting because you can hear a lot more of the environment at
the other end.
Tim
Gordon Henderson
Tim <nut...@kooky.org> wrote:
>> The G.722 codec that can be used with this service does appear to sound
>> better than a normal voip call.
>
>Indeed. But you can use G.722 with almost any SIP service which doesn't
>use asterisk or otherwise limit codecs. Asterisk can support G.722 but
>needs patching and a few things aren't quite right. I'm told this will
>be addressed in future versions.
I'm using G722 in Asterisk. (the back-port to 1.4) Works OK, but as you
say right now there are issues. The ones I've found are mostly to do with
transcoding and issues after putting a call on hold (and IAX - sticking
to SIP seems fine), but I need to do more tests. Have one beta-test
customer using it with polycom phones and they seem OK with it so-far.
>I have an SER SIP server I use for internal calls. G.722 is nice but it
>is disconcerting because you can hear a lot more of the environment at
>the other end.
It's like video phones - must remember to not pick nose :)
Gordon
Nick
all the phones. However my own phone is the only one that actually
receives. When I try to connect to the others I get the dialling sound
repeated and eventually IP Status Code 408.
Nick
> Nick wrote:
>> I have checked all phones are registered and they are. So I'm now
>> leaning towards the view that the service is pants, does anyone else
>> have any experience.
>
> It is a bit pants. Works great if you are a public IP address.
>
> No so good through NAT - has a tendency for 1 way audio.
>
I am using NAT but I can't see how it is a NAT issue. The phones are
registered but do not receive calls. I have my firewalls set up the same
, I'm forwarding the same ports.
> I'm not fussed by this because I have plenty of other SIP servers to use
> which work a lot better.
>
Yes I suppose this is the answer. I did look at Vaxalot but I don't
think SIP registration is included in the free account. Are there any
free ones.
Gordon Henderson
Nick <Ni...@spam.com> wrote:
>Tim wrote:
>> Nick wrote:
>>> I have checked all phones are registered and they are. So I'm now
>>> leaning towards the view that the service is pants, does anyone else
>>> have any experience.
>>
>> It is a bit pants. Works great if you are a public IP address.
>>
>> No so good through NAT - has a tendency for 1 way audio.
>>
>
>I am using NAT but I can't see how it is a NAT issue. The phones are
>registered but do not receive calls. I have my firewalls set up the same
>, I'm forwarding the same ports.
NAT is the biggest PITA with SIP. There are many ways round it, but
they're all "ways round it", alas.
You may be better off NOT port-forwarding and using a STUN server. Also
make sure your firewall/router/modem device does not have a SIP ALG
enabled.
>> I'm not fussed by this because I have plenty of other SIP servers to use
>> which work a lot better.
>>
>Yes I suppose this is the answer. I did look at Vaxalot but I don't
>think SIP registration is included in the free account. Are there any
>free ones.
So what you're after is basically a free, private "intercom" type of
thing? Why not sign up for a sipgate account for each extension - then
just dial the internal sipgate numbers..
I don't think you even have to pay them to just have a basic account.
Or run your own little asterisk box...
Gordon
Nick
>
> NAT is the biggest PITA with SIP. There are many ways round it, but
> they're all "ways round it", alas.
>
> You may be better off NOT port-forwarding and using a STUN server. Also
> make sure your firewall/router/modem device does not have a SIP ALG
> enabled.
>
I'm not quite happy with that answer but I'm not clever enough to
contradict you. It is a kind of interesting comment but I have other
things I should be learning even though this question been nagging at me
for about a year.
>>> I'm not fussed by this because I have plenty of other SIP servers to use
>>> which work a lot better.
>>>
>> Yes I suppose this is the answer. I did look at Vaxalot but I don't
>> think SIP registration is included in the free account. Are there any
>> free ones.
>
> So what you're after is basically a free, private "intercom" type of
> thing? Why not sign up for a sipgate account for each extension - then
> just dial the internal sipgate numbers..
>
> I don't think you even have to pay them to just have a basic account.
>
Well I'm just playing really. I had the phone and it worked so well I
got a few for relatives. I noticed the Gigaset.net account and it works
really well one way around but not the other so I wondered if there was
an easy fix.
I have a Sipgate account so I'll look at that. I don't really understand
what a SIP registration server does I had naively assumed it was a
lightweight directory type service that was only used to initiate the
initial connection. I guess the regular heartbeat style re-registrations
put to much load on a server to make it free.
Thanks for the help.
alexd
> Gordon Henderson wrote:
>
>>
>> NAT is the biggest PITA with SIP. There are many ways round it, but
>> they're all "ways round it", alas.
> I'm not quite happy with that answer but I'm not clever enough to
> contradict you.
Try his suggestion. See what happens. And roll on IPv6, so we can say
goodbye and good riddance to NAT!
>>> I did look at Vaxalot
Vaxalot sucks ;-)
> I have a Sipgate account so I'll look at that. I don't really understand
> what a SIP registration server does I had naively assumed it was a
> lightweight directory type service that was only used to initiate the
> initial connection. I guess the regular heartbeat style re-registrations
> put to much load on a server to make it free.
The above generally only becomes an issue when NAT is involved. SIP is
theoretically peer to peer, but when you throw NAT in, the endpoints can't
see each other, so need a server ['proxy', 'back-to-back user agent'] to
talk through. Add in buggy NAT and SIP ALG implementations and you've got a
whole host of reasons for hilarity.
Sipgate is one option, that gives you a PSTN number for each phone as a side
effect. pbxes.com is another, although I'm not sure about how their pricing
structure would work out for you.
--
<http://ale.cx/> (AIM:troffasky) (UnSoEs...@ale.cx)
13:26:23 up 12 days, 16:00, 2 users, load average: 0.19, 0.18, 0.14
A few flakes working together can unleash an avalanche of destruction
Gordon Henderson
Nick <Ni...@spam.com> wrote:
>Gordon Henderson wrote:
>
>>
>> NAT is the biggest PITA with SIP. There are many ways round it, but
>> they're all "ways round it", alas.
>>
>> You may be better off NOT port-forwarding and using a STUN server. Also
>> make sure your firewall/router/modem device does not have a SIP ALG
>> enabled.
>
>I'm not quite happy with that answer but I'm not clever enough to
>contradict you. It is a kind of interesting comment but I have other
>things I should be learning even though this question been nagging at me
>for about a year.
NAT is a big issue with SIP.
If you port-forward, you can only effectively have one device on the
inside. You may be able to have multiple accounts on that device though.
However, you need to tell the device what it's external IP address is.
The reason for this is that SIP encapsulates the endpoints IP address
inside the data/command packets it sends out.
So by default, if the phone is behind NAT, on (eg.) 192.168.1.10, then
it will encode that IP address inside the command packets. The recieving
side will take that IP address and use it to send audio back. Since
you can't get to that IP address except from behind your own NAT, it
won't work. They may be able to hear you, but you won't be able to hear
them. One way audio - and how many times have we read that about VoIP ...
There are several ways forward - one is telling the phone what it's
external IP address is, so it uses that IP address in the command packets
it sends out. In this scenario you usually need to port-forward on the
router too.
Another way is by using STUN. STUN is just a service out on the 'net
that allows a device to find out it's external IP address and how
port re-mapping happens. The phone can then use this information when
contacting a remote service.
Another way is by the remote service using a SIP proxy that can do "deep
packet inspection" and re-write the data packets so that it re-writes
the IP address to be the same as the IP address it gets the data from
before passing the packets on to the PBX.
Some modem/router/firewall devices have what's called a SIP ALG -
Application Layer Gateway. This tries to do the same "deep packet
inspection" on outgoing SIP data and do the re-writing, while at the
same time setting up more "memory" for the returning SIP/audio data.
I've yet to find a router which has a fully working SIP ALG.
Some routers even block SIP because they run their own VoiP
services. Fritz boxes, Draytek 'v' series and others. eg. BT HomeHubs.
>>>> I'm not fussed by this because I have plenty of other SIP servers to use
>>>> which work a lot better.
>>>>
>>> Yes I suppose this is the answer. I did look at Vaxalot but I don't
>>> think SIP registration is included in the free account. Are there any
>>> free ones.
>>
>> So what you're after is basically a free, private "intercom" type of
>> thing? Why not sign up for a sipgate account for each extension - then
>> just dial the internal sipgate numbers..
>>
>> I don't think you even have to pay them to just have a basic account.
>
>Well I'm just playing really. I had the phone and it worked so well I
>got a few for relatives. I noticed the Gigaset.net account and it works
>really well one way around but not the other so I wondered if there was
>an easy fix.
>
>I have a Sipgate account so I'll look at that. I don't really understand
>what a SIP registration server does I had naively assumed it was a
>lightweight directory type service that was only used to initiate the
>initial connection. I guess the regular heartbeat style re-registrations
>put to much load on a server to make it free.
A "lightweight directory type service" is good enough :)
It's either that, or you dial by IP address...
Gordon
Gordon Henderson
>Nick wrote:
>
>> Gordon Henderson wrote:
>>
>>>
>>> NAT is the biggest PITA with SIP. There are many ways round it, but
>>> they're all "ways round it", alas.
>
>> I'm not quite happy with that answer but I'm not clever enough to
>> contradict you.
>
>Try his suggestion. See what happens. And roll on IPv6, so we can say
>goodbye and good riddance to NAT!
And "Hello" to (lack of) firewall problems because by then everyone
will have forgotten what firewalls are, having been "protected" by NAT
for all this time...
Gordon