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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.

Sipgate : Signalling issues



 
 
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  #1  
Old February 21st 06, 11:06 PM posted to uk.telecom.voip
Andrew (G0RVM)
external usenet poster
 
Posts: 17
Default Sipgate : Signalling issues

I'm getting increasingly annoyed with Sipgate just now due to the many
outgoing PSTN calls that fail to connect. Quite often there is a long
delay between my INVITE and their 200 OK (40+ secs) together with the
establishment of bidirectional media streams.

These issues seem to have started this year and are very frustrating. Around
only 40% of the calls work. More frustratingly sometimes the PSTN end has
answered the call and is streaming media but the SIP signalling from
Sipgate that would communicate call establishment is not seen at my end.

Is anyone else using Sipgate for PSTN calls suffering from similar issues?


Also, and related to the above, can anyone clarify the purpose of a SIP '183
SESSION PROGRESS' message and when it should be sent? Can it replace a
'200 OK' under some circumstances? I ask, as it would appear Sipgate are
sending this when I would expect a '200 OK'. It contains the info I would
expect in a '200 OK' too.

Thanks,
Andrew

  #2  
Old February 22nd 06, 12:05 AM posted to uk.telecom.voip
Andrew \(G0RVM\)
external usenet poster
 
Posts: 4
Default Sipgate : Signalling issues

In this case I don't beleive its anything to do with my Intertex router.
I'm capturing/monitoring packets on the outside of the router, directly
before the cable modem.

Would be interested to know the router you had and to which one you moved?

Thanks,
Andrew

"Jono" wrote in message
k...
Andrew (G0RVM) laid this down on his screen :
I'm getting increasingly annoyed with Sipgate just now due to the many
outgoing PSTN calls that fail to connect. Quite often there is a long
delay between my INVITE and their 200 OK (40+ secs) together with the
establishment of bidirectional media streams.

These issues seem to have started this year and are very frustrating.
Around
only 40% of the calls work. More frustratingly sometimes the PSTN end
has
answered the call and is streaming media but the SIP signalling from
Sipgate that would communicate call establishment is not seen at my end.
Is anyone else using Sipgate for PSTN calls suffering from similar
issues?


Also, and related to the above, can anyone clarify the purpose of a SIP
'183
SESSION PROGRESS' message and when it should be sent? Can it replace a
'200 OK' under some circumstances? I ask, as it would appear Sipgate are
sending this when I would expect a '200 OK'. It contains the info I
would
expect in a '200 OK' too.

Thanks,
Andrew


I was experiencing exactly what you describe..............until I got a
better router. Virtually all the problems I have ever experience have been
at my end, not the provider's.




  #3  
Old February 22nd 06, 07:59 AM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 17
Default Sipgate : Signalling issues

Andrew (G0RVM) wrote:



"Jono" wrote in message
k...
Andrew (G0RVM) laid this down on his screen :
I'm getting increasingly annoyed with Sipgate just now due to the many
outgoing PSTN calls that fail to connect. Quite often there is a long
delay between my INVITE and their 200 OK (40+ secs) together with the
establishment of bidirectional media streams.

These issues seem to have started this year and are very frustrating.
Around
only 40% of the calls work. More frustratingly sometimes the PSTN end
has
answered the call and is streaming media but the SIP signalling from
Sipgate that would communicate call establishment is not seen at my end.
Is anyone else using Sipgate for PSTN calls suffering from similar
issues?


Also, and related to the above, can anyone clarify the purpose of a SIP
'183
SESSION PROGRESS' message and when it should be sent? Can it replace a
'200 OK' under some circumstances? I ask, as it would appear Sipgate
are
sending this when I would expect a '200 OK'. It contains the info I
would
expect in a '200 OK' too.

Thanks,
Andrew


I was experiencing exactly what you describe..............until I got a
better router. Virtually all the problems I have ever experience have
been at my end, not the provider's.


In this case I don't beleive its anything to do with my Intertex router.
I'm capturing/monitoring packets on the outside of the router, directly
before the cable modem.

Would be interested to know the router you had and to which one you moved?

Thanks,
Andrew


Originally had an Origo router, cheapo @ £20

Now got Linksys WRT54G with DD-WRT firmware.

  #4  
Old February 22nd 06, 10:25 AM posted to uk.telecom.voip
Thomas Kenyon
external usenet poster
 
Posts: 148
Default Sipgate : Signalling issues

Roger Barrett wrote:

Andrew
Im using the Linksys and getting exactly same problem - up to 45 second to
connect on a call - seems to have gotten a lot worse in the past week.
Roger.


Weird not getting that here. Mind you, I'm sure you knew I'd say that.
  #5  
Old February 22nd 06, 10:26 AM posted to uk.telecom.voip
Roger Barrett
external usenet poster
 
Posts: 39
Default Sipgate : Signalling issues


"Jono" wrote in message
. uk...
Andrew (G0RVM) wrote:



"Jono" wrote in message
k...
Andrew (G0RVM) laid this down on his screen :
I'm getting increasingly annoyed with Sipgate just now due to the many
outgoing PSTN calls that fail to connect. Quite often there is a long
delay between my INVITE and their 200 OK (40+ secs) together with the
establishment of bidirectional media streams.

These issues seem to have started this year and are very frustrating.
Around
only 40% of the calls work. More frustratingly sometimes the PSTN end
has
answered the call and is streaming media but the SIP signalling from
Sipgate that would communicate call establishment is not seen at my
end.
Is anyone else using Sipgate for PSTN calls suffering from similar
issues?


Also, and related to the above, can anyone clarify the purpose of a SIP
'183
SESSION PROGRESS' message and when it should be sent? Can it replace a
'200 OK' under some circumstances? I ask, as it would appear Sipgate
are
sending this when I would expect a '200 OK'. It contains the info I
would
expect in a '200 OK' too.

Thanks,
Andrew

I was experiencing exactly what you describe..............until I got a
better router. Virtually all the problems I have ever experience have
been at my end, not the provider's.


In this case I don't beleive its anything to do with my Intertex router.
I'm capturing/monitoring packets on the outside of the router, directly
before the cable modem.

Would be interested to know the router you had and to which one you
moved?

Thanks,
Andrew


Originally had an Origo router, cheapo @ £20

Now got Linksys WRT54G with DD-WRT firmware.

Andrew
Im using the Linksys and getting exactly same problem - up to 45 second to
connect on a call - seems to have gotten a lot worse in the past week.
Roger.


  #6  
Old February 22nd 06, 10:27 AM posted to uk.telecom.voip
Keith Lawrence
external usenet poster
 
Posts: 10
Default Sipgate : Signalling issues

"Andrew (G0RVM)" wrote...

I'm getting increasingly annoyed with Sipgate just now due to the many
outgoing PSTN calls that fail to connect. Quite often there is a long
delay between my INVITE and their 200 OK (40+ secs) together with the
establishment of bidirectional media streams.

These issues seem to have started this year and are very frustrating.

Around
only 40% of the calls work. More frustratingly sometimes the PSTN end has
answered the call and is streaming media but the SIP signalling from
Sipgate that would communicate call establishment is not seen at my end.

Is anyone else using Sipgate for PSTN calls suffering from similar issues?


Seeing the same thing Andrew, although I haven't traced it at packet level.
It's definately Sipgate, when Sipgate fails (frequently, it's my 'junk'
line) the second line from Gradwell on the same Sipura SPA 2000 connects
inbound and outbound perfectly.

Keith L


  #7  
Old February 22nd 06, 12:46 PM posted to uk.telecom.voip
Alex
external usenet poster
 
Posts: 31
Default Sipgate : Signalling issues


"Andrew (G0RVM)" wrote in message
...
I'm getting increasingly annoyed with Sipgate just now due to the many
outgoing PSTN calls that fail to connect. Quite often there is a long
delay between my INVITE and their 200 OK (40+ secs) together with the
establishment of bidirectional media streams.

These issues seem to have started this year and are very frustrating.
Around
only 40% of the calls work. More frustratingly sometimes the PSTN end has
answered the call and is streaming media but the SIP signalling from
Sipgate that would communicate call establishment is not seen at my end.

Is anyone else using Sipgate for PSTN calls suffering from similar issues?


Also, and related to the above, can anyone clarify the purpose of a SIP
'183
SESSION PROGRESS' message and when it should be sent? Can it replace a
'200 OK' under some circumstances? I ask, as it would appear Sipgate are
sending this when I would expect a '200 OK'. It contains the info I would
expect in a '200 OK' too.

Thanks,
Andrew


Same here. Tried to call a mobile yesterday, took about 5 attempts. Then
when it did connect to the mobiles voicemail (they didn't pick up) it
dropped the call before the beep so I couldn't leave a message. There have
been quite a few dropped calls and failed outbound calls lately. Plus quite
alot of the time I don't get a ringing tone when I'm calling someone, its
just silent until they pick up. I know that this is generated my end,
however it only works intermittently on both my Grandstream ATA and Draytek
voip router, which leads me to believe sipgate aren't sending out a signal
when they should.

My Grandstream has sat without configuration change for about a year now,
and my Draytek for about 6 months- it used to work perfectly, its only as
late that I've been having problems. I was kind of leaving them to it hoping
that it'd get sorted out before I try another provider, as I have two nifty
memorable Sipgate PSTN numbers which everyone knows, changing it would be a
pig, plus they were good up until not too long ago...

Alex


  #8  
Old February 22nd 06, 01:22 PM posted to uk.telecom.voip
Andrew (G0RVM)
external usenet poster
 
Posts: 17
Default Sipgate : Signalling issues

Alex wrote:


"Andrew (G0RVM)" wrote in message
...
I'm getting increasingly annoyed with Sipgate just now due to the many
outgoing PSTN calls that fail to connect. Quite often there is a long
delay between my INVITE and their 200 OK (40+ secs) together with the
establishment of bidirectional media streams.

These issues seem to have started this year and are very frustrating.
Around
only 40% of the calls work. More frustratingly sometimes the PSTN end
has answered the call and is streaming media but the SIP signalling from
Sipgate that would communicate call establishment is not seen at my end.

Is anyone else using Sipgate for PSTN calls suffering from similar
issues?


Also, and related to the above, can anyone clarify the purpose of a SIP
'183
SESSION PROGRESS' message and when it should be sent? Can it replace a
'200 OK' under some circumstances? I ask, as it would appear Sipgate are
sending this when I would expect a '200 OK'. It contains the info I
would expect in a '200 OK' too.

Thanks,
Andrew


Same here. Tried to call a mobile yesterday, took about 5 attempts. Then
when it did connect to the mobiles voicemail (they didn't pick up) it
dropped the call before the beep so I couldn't leave a message. There have
been quite a few dropped calls and failed outbound calls lately. Plus
quite alot of the time I don't get a ringing tone when I'm calling
someone, its just silent until they pick up. I know that this is generated
my end, however it only works intermittently on both my Grandstream ATA
and Draytek voip router, which leads me to believe sipgate aren't sending
out a signal when they should.

My Grandstream has sat without configuration change for about a year now,
and my Draytek for about 6 months- it used to work perfectly, its only as
late that I've been having problems. I was kind of leaving them to it
hoping that it'd get sorted out before I try another provider, as I have
two nifty memorable Sipgate PSTN numbers which everyone knows, changing it
would be a pig, plus they were good up until not too long ago...

Alex


Alex,

Thanks for the info. I am going to start another thread in the next few
days once I have accumulated some more stats. But a brief note here won't
go amiss.

I have now captured several calls from/to my Sigate account using the PSTN
and to another account I have with Sipgate, thus with the latter, staying
in the IP universe.

All calls have been captured using ethereal on the outside of my
firewall/router - in this way I get to see exactly what is being exchanged
with Sipgate.

The first thing I realised is that Ring tones may be generated locally at
the client OR remotely by the network. In the former case a '180 RINGING'
is received. Alternatvely, is seems possible to omit this message and pass
the Ring tone in the RTP media stream. In that case a '182 Session
Progress' message is received. (Someone correct me if I've got that wrong)

Now I routinely suffer from the same problem as you - No Ring tone - the
called person just answers the phone. This is caused by, in my case, the
Grandstream HT-286 not getting a '180 RINGING' message from Sipgate. (when
it does - it rings) My HT-286 does not seem to process the '182 SESSION
PROGRESS' message correctly, probably understanding it as a '100 TRYING'.
Therefore it does not realise the Ring tones are being sent and start
replaying the incoming media stream to the client device.

I have identified a number of other issues too related to signalling
latencies and also Sipgate just not generating required signalling.

All this is not random either, it occurs worryingly often. Now, I'm pretty
sure I'm not barking up the wrong tree, but just to make sure I'm going to
start another thread (mentioned earlier) to see if other people make of my
tests, results and conclusions.

Regards,
Andrew
  #9  
Old February 22nd 06, 01:23 PM posted to uk.telecom.voip
Thomas Kenyon
external usenet poster
 
Posts: 148
Default Sipgate : Signalling issues

Andrew (G0RVM) wrote:

Now I routinely suffer from the same problem as you - No Ring tone - the
called person just answers the phone.

I've had this when using international calling cards (from a payphone).
  #10  
Old February 22nd 06, 04:17 PM posted to uk.telecom.voip
{{{{{Welcome}}}}}
external usenet poster
 
Posts: 908
Default Sipgate : Signalling issues

Thus spaketh Thomas Kenyon:
Roger Barrett wrote:

Andrew
Im using the Linksys and getting exactly same problem - up to 45
second to connect on a call - seems to have gotten a lot worse in
the past week. Roger.


Weird not getting that here. Mind you, I'm sure you knew I'd say that.


I'm using a LinkSys PAP2 connected to a LinkSys router, and have no problems.


 




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