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uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed.

Which codec for Betamax \ Dellmont in Siemens A580


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  #1  
Old December 6th 18, 03:21 PM posted to uk.telecom.voip
Ken[_2_]
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Posts: 2
Default Which codec for Betamax \ Dellmont in Siemens A580

I'm using a Siemens A580 IP for my VOIP calls. Using Discount Voip for my
outgoing calls. It works OK but I can certainly tell an audio difference
between that and say Voipfone, which is what I am using for incoming calls.

I was just wondering what codec selection other folk were using for the
various Betamax services. I'm on a 150 mbps FTTP connection.
  #2  
Old December 6th 18, 03:31 PM posted to uk.telecom.voip
Graham.[_3_]
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Posts: 289
Default Which codec for Betamax \ Dellmont in Siemens A580

I'm using a Siemens A580 IP for my VOIP calls. Using Discount Voip for my
outgoing calls. It works OK but I can certainly tell an audio difference
between that and say Voipfone, which is what I am using for incoming calls.

I was just wondering what codec selection other folk were using for the
various Betamax services. I'm on a 150 mbps FTTP connection.


G.711 A-Law is pretty much standard.

I think I once used GSM to see how many simultaneous calls I could
squeeze down my ADSL pipe, but they sounded like mobiles
(unsurprisingly)

--
Graham.

%Profound_observation%
  #3  
Old December 9th 18, 12:58 AM posted to uk.telecom.voip
Pancho
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Posts: 8
Default Which codec for Betamax \ Dellmont in Siemens A580

On 06/12/2018 14:21, Ken wrote:
I'm using a Siemens A580 IP for my VOIP calls. Using Discount Voip for my
outgoing calls. It works OK but I can certainly tell an audio difference
between that and say Voipfone, which is what I am using for incoming calls.

I was just wondering what codec selection other folk were using for the
various Betamax services. I'm on a 150 mbps FTTP connection.


I also noticed inferior audio from Betamax, this time compared to
Sipgate. In particular this was on voip to voip connections.

I have always felt that SIP was an inferior protocol in that
implementations often don't negotiate point to point connections, my
suspicion was that Betamax calls always routed via Belgium or
Netherlands and that this added latency which
degraded audio. I've no idea how reasonable my suspicion is or if this
is a problem with SIP or just the SIP providers like Betamax.

It just seem to be that SIP is a bit like email, a protocol that emerged
early on in the internet history and then became frozen despite obvious
improvements that could be made.


  #4  
Old December 9th 18, 12:18 PM posted to uk.telecom.voip
David Woolley
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Posts: 99
Default Which codec for Betamax \ Dellmont in Siemens A580

On 08/12/2018 23:58, Pancho wrote:
I have always felt that SIP was an inferior protocol in that
implementations often don't negotiate point to point connections, my
suspicion was that Betamax calls always routed via Belgium or
Netherlands and that this added latency which
degraded audio. I've no idea how reasonable my suspicion is or if this
is a problem with SIP or just the SIP providers like Betamax.


The reasons for not negotiating direct media are a mix of commercial,
and security ones.

Firstly, like SMTP, SIP can do point to point signalling set up. SMTP
isn't used like that for commercial reasons (low end ISPs don't offer
it) and security reasons (people who try to initiate point to point SMTP
are likely to be assumed to be spammers, as many ISPs only accept
inbound SMTP from other big ISPs.

SIP can also do point to point media, with centralised signalling.
There are some technical difficulties with NAT, but there are additions
to the SDP and RTP protocols to deal with those. If you want to use
phone numbers as your only address, you do need some centralised
support, although SIP itself is not limited to phone numbers and can use
arbitrary [email protected] addresses.

There has been a lot of recent development to SDP to aid in NAT traversal.

I believe the typical use of SIP in WebRTC involves point to point
signalling set up, although I'm not sure that it allows a different
media path.

All of the SIP hard and soft phones I've used support point to point
signalling, at least in a non-NAT environment. Some of them support
direct media, although the version of X-Lite I tried, had a broken
Re-INVITE implementation, so could not be put into direct media mid-call.

I think the real market for SIP hard phones is the corporate one, where
direct media may well be used on the internal network (although the need
to monitor for DTMF may frustrate this - an older feature INFO, can
avoid that issue).
  #5  
Old December 9th 18, 05:59 PM posted to uk.telecom.voip
David Higton
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Posts: 8
Default Which codec for Betamax \ Dellmont in Siemens A580

In message
Pancho wrote:

I have always felt that SIP was an inferior protocol in that
implementations often don't negotiate point to point connections, my
suspicion was that Betamax calls always routed via Belgium or
Netherlands and that this added latency which
degraded audio. I've no idea how reasonable my suspicion is or if this
is a problem with SIP or just the SIP providers like Betamax.


It's absolutely not the problem of SIP. The protocol itself /does/
support point-to-point media. Whether any given connection will
use point-to-point media is entirely another question, and not due
to any shortcoming of the protocol itself.

Dave
  #6  
Old December 10th 18, 01:20 AM posted to uk.telecom.voip
Pancho
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Posts: 8
Default Which codec for Betamax \ Dellmont in Siemens A580

On 09/12/2018 11:18, David Woolley wrote:
On 08/12/2018 23:58, Pancho wrote:
I have always felt that SIP was an inferior protocol in that
implementations often don't negotiate point to point connections, my
suspicion was that Betamax calls always routed via Belgium or
Netherlands and that this added latency which
degraded audio. I've no idea how reasonable my suspicion is or if this
is a problem with SIP or just the SIP providers like Betamax.


The reasons for not negotiating direct media are a mix of commercial,
and security ones.

Firstly, like SMTP, SIP can do point to point signalling set up.* SMTP
isn't used like that for commercial reasons (low end ISPs don't offer
it) and security reasons (people who try to initiate point to point SMTP
are likely to be assumed to be spammers, as many ISPs only accept
inbound SMTP from other big ISPs.


My criticism with email is related to encryption, security, identity not
that it is not P2P. Whatsapp app and such like can manage it but for
whatever reason open standard secure email formats were never widely used.

Wouldn't it be great if banks and other services used a secure push
message service, like encrypted email, rather than me having to go
through a tedious log on to retrieve statements.

SIP can also do point to point media, with centralised signalling. There
are some technical difficulties with NAT, but there are additions to the
SDP and RTP protocols to deal with those.* If you want to use phone
numbers as your only address, you do need some centralised support,
although SIP itself is not limited to phone numbers and can use
arbitrary [email protected] addresses.

There has been a lot of recent development to SDP to aid in NAT traversal.



As recently as a couple of weeks ago I had to debug my Mum's Gigaset
voip phone set up and reboot her router after something, possibly the
SIP ALG failed.

Recent developments sounds a bit lame given how long the NAT problem has
existed. Perhaps they will come up with a really good solution just in
time for universal IPV6 adoption to make it redundant. Sorry I sound
bitter, its not with you. Your response was helpful and I understand
where you are coming from, I just get frustrated.

I believe the typical use of SIP in WebRTC involves point to point
signalling set up, although I'm not sure that it allows a different
media path.

All of the SIP hard and soft phones I've used support point to point
signalling, at least in a non-NAT environment.* Some of them support
direct media, although the version of X-Lite I tried, had a broken
Re-INVITE implementation, so could not be put into direct media mid-call.


Yep I tried to setup Re-Invite (is it called direct media now) as a
pre-requisite for using Asterisk at home but unfortunately failed. I
half remember that the reason for this was Sipgate (my voip service) not
supporting re-invite at that time. I think Sipgate may have fixed this
in the interim years.

I think the real market for SIP hard phones is the corporate one, where
direct media may well be used on the internal network (although the need
to monitor for DTMF may frustrate this - an older feature INFO, can
avoid that issue).


AIUI most of this group are residential users of SIP "hard" phones. If I
wanted to uses a PC solution, like X-Lite, I would use Skype in
preference. I do use soft SIP phone Zoiper on my mobile phone to pick up
calls to my house phone, it can be problematic.

I also understand that in theory my Siemens Gigaset may be configurable
to allow direct connections, but when I have tried over the past 10+
years I have failed to get this to work.

As I said sorry to sound bitter. It is just that VOIP promises so much
but configuring SDP (SIP/RTP) has proved to be very problematic for me
over the many years I have used it.
  #7  
Old December 10th 18, 11:08 AM posted to uk.telecom.voip
Theo[_2_]
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Posts: 68
Default Which codec for Betamax \ Dellmont in Siemens A580

Pancho wrote:
I also understand that in theory my Siemens Gigaset may be configurable
to allow direct connections, but when I have tried over the past 10+
years I have failed to get this to work.

As I said sorry to sound bitter. It is just that VOIP promises so much
but configuring SDP (SIP/RTP) has proved to be very problematic for me
over the many years I have used it.


I think an analogy might be a bit like if we're expecting a printer to run
its own public print server to allow direct connections to/from internet
clients. Yes you can do it, but in a company typically it will be slaved
off a bigger server. In the case of consumer printers these days, for
mobile printing that server is run by Apple or Google.

A lot of IP phones don't have a whole lot of compute, and expecting them to
communicate to anyone over the Big Bad Internet might be a bit much. In a
company the IP phones would be slaved off a bigger SIP server that can then,
subject to policy, do direct connection to another SIP server. It's a lot
easier for the company to manage the SIP server than the configuration of
thousands of desk phones (which are now getting their config push from the
central server).

This is still peer-to-peer, but P2P in the mode of old-fashioned SMTP where
each company would run its own SMTP server (instead of the current
Gmail/Outlook/etc world). Old-fashioned SMTP was never completely P2P
except perhaps when microcomputers were only dumb terminals and so didn't
count as nodes.

So maybe what you need is a minimalist 'central' SIP server, eg Asterisk on
a Raspberry Pi. That's your VAX, and your SIP phones are your dumb terminals.

Theo
  #8  
Old December 10th 18, 05:55 PM posted to uk.telecom.voip
Ken[_2_]
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Posts: 2
Default Which codec for Betamax \ Dellmont in Siemens A580

On Sun, 09 Dec 2018 16:59:34 GMT, David Higton wrote:

It's absolutely not the problem of SIP. The protocol itself /does/
support point-to-point media. Whether any given connection will
use point-to-point media is entirely another question, and not due
to any shortcoming of the protocol itself.

Dave


Have been away for a couple of days so just wanted to say thanks for the codec
suggestion and the discussion. Like most others I'm behind NAT so the
discussion is of great interest.
  #9  
Old Yesterday, 01:31 AM posted to uk.telecom.voip
Pancho
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Posts: 8
Default Which codec for Betamax \ Dellmont in Siemens A580

On 10/12/2018 10:08, Theo wrote:


So maybe what you need is a minimalist 'central' SIP server, eg Asterisk on
a Raspberry Pi. That's your VAX, and your SIP phones are your dumb terminals.

A bit sad that you expect me to know what a VAX is. Even sadder that I do.
  #10  
Old Yesterday, 02:03 PM posted to uk.telecom.voip
MissRiaElaine[_2_]
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Posts: 136
Default Which codec for Betamax \ Dellmont in Siemens A580

On 12/12/2018 00:31, Pancho wrote:
On 10/12/2018 10:08, Theo wrote:


So maybe what you need is a minimalist 'central' SIP server, eg
Asterisk on
a Raspberry Pi.* That's your VAX, and your SIP phones are your dumb
terminals.

A bit sad that you expect me to know what a VAX is. Even sadder that I do.


It's a vacuum cleaner, isn't it..?

--
Ria in Aberdeen

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