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VoIP Beta Trial - outbound calls terminated early?

vof
Grafter
Posts: 43
Registered: 04-08-2007

VoIP Beta Trial - outbound calls terminated early?

I've used the service for 7 outbound calls this morning. The first two were fine. The next 5 have had problems or I suspect have had problems.
In particular, one call ended suddenly in mid-conversation.
At least three other calls were to leave messages on answering machines. In these cases, the call length recorded in the Calls list is much shorter than the actual time displayed on the handset.
Curiously, four of the five problem calls are logged as 36 seconds - coincidence?
I am still suspicious about the extremely short registration interval in use by my ATA (25 seconds). Can PN ask Gradwell if this is expected/normal behaviour?
Finally, how do PN want incidents like this to be reported? Through this forum or via tickets?
11 REPLIES
RichSmol
Grafter
Posts: 709
Registered: 29-10-2007

Re: VoIP Beta Trial - outbound calls terminated early?

No no, this is definately the right place for these types of issues so others can contribute and add their names to problems if they're experiencing the same problems. Jake (Gradwell rep) will be along shortly to help me answer these questions.
jakejohnson
Dabbler
Posts: 22
Registered: 06-11-2007

Re: VoIP Beta Trial - outbound calls terminated early?

Hi vof,
To help you out further it would be great if you could tell us your extension number and also anything thats on your network such as routers, devices, switches, computers, etc.
Generally we tend to link call drops back to firewall rules. So if you can tell us ythose as well that'd be great.
As for your registration time of your ATA, it is unusual. We prefer the registration time of an hour with our systems.
Thanks,
Jake
vof
Grafter
Posts: 43
Registered: 04-08-2007

Re: VoIP Beta Trial - outbound calls terminated early?

@Jake
All my network devices are linked via an HP Procurve 408 switch:
2 Windows PCs
Linux server
HP LJ5M printer
Sipura SPA-1001 ATA (hardware version 2.0.1(99e2) with firmware version 2.0.13(SEg))
3Com 3CRWDR101A-75 Router
Router configuration:
Firewall level set to Medium (not sure what that means in detail)
No rules other than port forwarding:
5060 (UDP)
5061 (UDP)
5004 (UDP)
10000 (UDP)
3478 (UDP)
3478 (TCP)
5082 (UDP) [in case it was relevant to the Beta Trial]
The ATA has fixed IP address and all traffic is mapped to the highest priority EF (Expedited Forwarding) traffic stream in the router.
I have always been surprised at how much data the ATA generates even when quiescent - currently about 200 bytes/sec - is this typical/normal?
The other service on the ATA is Sipgate. Both services have identical configurations (apart from the SIP ports and service related addresses/ports). Both have a registration interval configured as 3600 seconds - registration occurs every 25 seconds on PN, every 600 seconds on Sipgate.
I'll email my extension to you. Let me know if there is anything else you need to know.
vof
Simon_M
Grafter
Posts: 684
Registered: 05-04-2007

Re: VoIP Beta Trial - outbound calls terminated early?

[O.T]
Jake
This is an open forum. Unless your posted email address is a sacrificial one for the purposes of this trial, I don't think it's a good idea to post it in plain text.
The Private Message (PM) system is safer & can be used to exchange email addresses if you need to take a conversation to email.
jakejohnson
Dabbler
Posts: 22
Registered: 06-11-2007

Re: VoIP Beta Trial - outbound calls terminated early?

Hi vof,
Can you log into the ATA using this url http://ipaddress/admin/advanced, then click the SIP tab.
In here you will find the RTP port Min and Max, which are the ports you will need to forward in your router. You can also find in Ext1 and Ext2 the SIP port to confirm they are on the correct ones for the port forwarding you have in place. In the Ext1 or Ext2 can you ensure that your extension is set as "Register Expires: 3600"
Thank,
Jake
vof
Grafter
Posts: 43
Registered: 04-08-2007

Re: VoIP Beta Trial - outbound calls terminated early?

@Jake:
RTP Port Min was set to 16384, RTP Port Max to 16482. I had not originally set up any forwarding for ports in this range.
The router firewall port field is too short to hold this range of five digit port numbers: 16384-16482 (!) - I assume it is a range - so I changed RTP Port Max to 16391 giving an 8 port range which I entered as eight individual port mappings. Tried both UDP, then TCP, without any effect.
Line 1 is set to use SIP Port 5060 for the PlusNet VoIP Beta Trial. Line 2 is set to use SIP Port 5061 for Sipgate. Both 5060/UDP and 5061/UDP are forwarded in router (and have been since I first used VoIP). Both lines have Register Expires: 3600 - neither seems effective with PlusNet registration happening every 25 seconds and Sipgate every 600 seconds. I can't recall whether this Sipgate interval has always been 600 seconds.
Four outbound calls today have been fine (I think).
My feedback on general call quality - very good and no obvious difference from BT, though I do agree about the pips in the speaking clock which generally has a poorer, synthesized feel to it.
jakejohnson
Dabbler
Posts: 22
Registered: 06-11-2007

Re: VoIP Beta Trial - outbound calls terminated early?

Hi vof,
For your registration problem, we are running a trace on your extension and will get abck to you when we know more.
As for your port issue, it should be a 98 port range. You can set this quite low if need be.
And finally, the speaking clock. This is on a different server ot our VoIP platform as we don't want it using up all our resources! The server it is on is rather old and quite loaded so the quality will not be on par with VoIP calls, although we are making infrastructure changes at the moment so it may be refreshed some time soon.
Thanks,
Jake
edit/ We found from running the traces that the device is trying to register but it looks like the device is set to use STUN Test enable, or STUN enabled on the spa 1001. You will need to turn this off in the device.
Can you also make sure that the device is set under the extension tab to 'register: yes'
User-Agent: Sipura/SPA1001-2.0.13(SEg).
Warning: 399 spa "Full Cone NAT Detected".
Authorization: Digest
vof
Grafter
Posts: 43
Registered: 04-08-2007

Re: VoIP Beta Trial - outbound calls terminated early?

@Jake
Thanks for info and suggestions.
I had STUN and STUN Test enabled but have now disabled both. As a result, the PN VoIP line registration is now 3600 seconds (though Sipgate is still 600). Thanks! The Info tab on the device currently shows a blank in the 'Mapped SIP port' for the PN line which is actually using 5060. (The Sipgate line is reported as using Mapped SIP port 5061.) This may be a glitch in the Sipura firmware but I will keep an eye on it.
The STUN setting are device rather than line level settings so STUN has been disabled for Sipgate too. I have just done a simple check and outbound and inbound calls appear to still work on both lines. It is not always clear when a VoIP service requires a STUN server to be used so does the PN requirement not to use it limit which services I can use on the second line of the ATA?
Both lines/extensions have always been set to Register: yes.
Is the RTP port range forwarding still a requirement? What effect is not forwarding likely to have? How low is quite low for the RTP port range ;>) What do you suggest as the lowest base port for the range which seems to be 99 ports long not 98. And does RTP use UDP or TCP?
vof
jakejohnson
Dabbler
Posts: 22
Registered: 06-11-2007

Re: VoIP Beta Trial - outbound calls terminated early?

Quote from: vof
ow low is quite low for the RTP port range ;>) What do you suggest as the lowest base port for the range which seems to be 99 ports long not 98. And does RTP use UDP or TCP?

Hi vof,
Are able to use 8000-8098 or if that is still to large for your router to accept?
The RTP range is just to make sure that the audio streams reach the phone and dont cause any one way audio. All our traffic will be in UDP and not TCP.
Thanks,
Jake
walford58
Dabbler
Posts: 11
Registered: 03-12-2007

Re: VoIP Beta Trial - outbound calls terminated early?

Hi
I'm using a Linksys SPA9000 and just gone over to the trial.
Registration with the PN is fine, but I'm also getting renewal every 25 seconds.
The firewall is set to  route all the necessary ports, as had been working previously with the old PlusTalk (although the inbound calls were the problem for me).
Sipgate works fine.
Also I am getting reports of the conversation (outbound-dialled) being terminated without warning tones after about 30 seconds.
I have RTP ports 8000 to 16482 set as the range on the SIP tab and STUN is disabled - I found this was necessary for Sipgate and by putting a direct port mapping for the PN line enabled me to disable it.
Extension number is 8950109, if you could please check the trace on the registration, thanks.
I've not been able to check the call quality yet.
vof
Grafter
Posts: 43
Registered: 04-08-2007

Re: VoIP Beta Trial - outbound calls terminated early?

@Jake
Apologies for delay in getting back on this.
I spoke too soon - the registration interval is back at 25-ish seconds even though STUN is still disabled. The Info tab on the device still shows a blank in the 'Mapped SIP port' for the PN line which is actually using 5060. I have rebooted the ATA to check the effect is there across a reboot.
My router will only allow a two digit port number for the first and last port in a range so 8000-8098 is not accepted. It actually only allows a maximum of 5 characters in the field(!)
However, outbound calls still seem to work OK though I did have a sudden call disconnection this morning at exactly 2 minutes into call.
vof