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SIP phones - Inbound calls drop out after 32s on Vigor 2710N

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10 Aug 2011 21:28 #68992 by southendsupporter
Hi
I am using a Draytek 2710n to provide access to Entanet's SIP server for (at the moment) 3 Yaelink SIPT20 SIP handsets.
Everything is working EXCEPT that incoming calls drop out after 32secs.
Entanet support have suggested ensuring that UDP 10000 to 20000 and UDP/TCP 5060 are open on the firewall.
Under NAT>OpenPorts>EditOpenPorts I can see how to enter the ports and protocol but it would seem that I can only open them for 1 target IP address?
Do I therefor have to have a 'rule' for each IP address or is there a way to open the ports generally?
Also, they asked me to ensure that 'SIPALG' is disabled. I cannot find any reference to this in the management interface or the manual?

Lastly, I was told to configure the phones to use a STUN server with a STUN interval of 15 seconds. Whilst the Yaelink enables me to set a STUN server and enable NAT Traversal in order to use it, I cannot set the stun interval.
Can this be configured effectively in the router? The phones are connecting directly to to the customer's voip server via the router so the VoIP is essentially transparent to the router?

Hope you can help.

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11 Aug 2011 09:22 #68999 by nealuk
Replied by nealuk on topic Re: Opening Ports on Draytek 2710n
Firstly, I would suggest that in a multiple handset environment the sip registrations are on different ports.

e.g. based on the assumption that each handset has one sip account registered.
Handset 1
192.168.0.11
SIP Reg Port 5060

Handset 2
192.168.0.12
SIP Reg Port 5062

Handset 3
192.168.0.13
SIP Reg Port 5064

Amend the setting on the Yaelink SIPT20 SIP handsets so that they know they are using 5060 / 5062 / 5064 respectively.

Set open ports to forward TCP/UDP 5060-5061 to Handset 1, 5062-5063 to Handset 2 and 5064-5065 to Handset 3.

Secondly, extrapolating on this, taking experience from an installation with a dozen IP handsets, some of which register up to 4 sip accounts, (but again based on the assumption that in your situation each handset has one sip account registered) here's what I'd suggest:

Handset 1
192.168.0.11
Voice Packets Port Range 10000 - 10050

Handset 2
192.168.0.12
Voice Packets Port Range 10051 - 10101

Handset 3
192.168.0.11
Voice Packets Port Range 10102 - 10152

Amend the setting on the Yaelink SIPT20 SIP handsets so that they know they are using Minimum RTP Port of 10000 / 10051 / 10102 respectively.

Amend the setting on the Yaelink SIPT20 SIP handsets so that they know they are using Maximum RTP Port of 10050 / 10101 / 10152 respectively.

Set open ports to forward TCP/UDP 10000-10050 to Handset 1, 10051-10101 to Handset 2 and 10102-10152 to Handset 3.

Reboot everything and hopefully things should be improved. Regards, Neal

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11 Aug 2011 12:47 #69004 by southendsupporter
Replied by southendsupporter on topic Re: Opening Ports on Draytek 2710n

nealuk wrote: Firstly, I would suggest that in a multiple handset environment the sip registrations are on different ports.

e.g. based on the assumption that each handset has one sip account registered.
Handset 1
192.168.0.11
SIP Reg Port 5060

Handset 2
192.168.0.12
SIP Reg Port 5062

Handset 3
192.168.0.13
SIP Reg Port 5064

Amend the setting on the Yaelink SIPT20 SIP handsets so that they know they are using 5060 / 5062 / 5064 respectively.

Set open ports to forward TCP/UDP 5060-5061 to Handset 1, 5062-5063 to Handset 2 and 5064-5065 to Handset 3.

Secondly, extrapolating on this, taking experience from an installation with a dozen IP handsets, some of which register up to 4 sip accounts, (but again based on the assumption that in your situation each handset has one sip account registered) here's what I'd suggest:

Handset 1
192.168.0.11
Voice Packets Port Range 10000 - 10050

Handset 2
192.168.0.12
Voice Packets Port Range 10051 - 10101

Handset 3
192.168.0.11
Voice Packets Port Range 10102 - 10152

Amend the setting on the Yaelink SIPT20 SIP handsets so that they know they are using Minimum RTP Port of 10000 / 10051 / 10102 respectively.

Amend the setting on the Yaelink SIPT20 SIP handsets so that they know they are using Minimum RTP Port of 10050 / 10101 / 10152 respectively.

Set open ports to forward TCP/UDP 10000-10050 to Handset 1, 10051-10101 to Handset 2 and 10102-10152 to Handset 3.

Reboot everything and hopefully things should be improved. Regards, Neal


Neal, thanks for a very full response. Need to get on-site to do this and will let you know how I get on!

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11 Aug 2011 20:25 #69010 by southendsupporter
Replied by southendsupporter on topic SIP phones - Inbound calls drop out after 32s on Vigor 2710N
Hi Neal,
disappointed of haslemere! - Followed your guidance and the problem persists :(
The detail in your post suggests you are familiar wioth VoIP systems?

I am a new partner with Entanet and their support team has been very willing and active in trying to help me but somewhat unsuccessful!

I installed an xLite softphone on a laptop on the customers network. This works perfectly!
So, this suggests that the problem lies with a handset setting as the same router and network was being used?

BUT... I plugged one of the handsets into my own network and if I opened all UDP ports on my router, or indeed just 5060, it worked!!

This in fact points to the router!!

The only difference I can find is that my Netgear router has SIP ALG enabled (or to be more accutae it is NOT disabled).

I cannot find how to verify if the 2710n has a similar setting - it isn't immediaely obvious in the actual web interface and I cannot find it in the manual?

Any ideas?

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11 Aug 2011 23:09 #69011 by southendsupporter
Replied by southendsupporter on topic SIP phones - Inbound calls drop out after 32s on Vigor 2710N
OK - I enabled the SIP ALG on my netgear and the phone still works on my Network so presumably whether this is enabled or disabled on the router isn't relevant.

Or might this not be the case on a Vigor 2710N??

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12 Aug 2011 09:13 #69014 by southendsupporter
Replied by southendsupporter on topic Re: SIP phones - Inbound calls drop out after 32s on Vigor 2
I found another thread similar to my issue which suggested isuong sys sip_alg 0 via the telnet interface.

Tried this and also sys sip_alg 1

neither solved the problem :(

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