Sip Signal Does Not Arrive At The Server After Several Minutes

[iTime]

Hello VG support,

 

We are having a problem with a new SIP trunk to our IVR (VG) server.

It is HMP/VG environment with SIP provision from a vendor called, "Fusion."

Once VG service starts, calls can be made to the server all right, but after about 5 minutes of inactivity, VG would not receive any calls at all.

From Wireshark, we can see there is no SIP or RTP signal arriving at the server either.

We have contacted the vendor (Fusion) for help, but it seems to take long time to get response from them, so we would like to have an advice from you as well.

 

The config we used for the connection is as shown below. It is static IP with no authentication. "peer2.thevoice...com" in <RegServer> is what we were told to use by the vendor, and we tried to use different values for "RegClient" with our IP address (after "@") or just the phone number itself etc.

We also tried different numbers for "<Expires>" to make it to be 30, 1800 or 36000, but no use.

I have attached all the logs I could obtain including Wireshark log. You will be able to see that there is no sign of incoming call around 11:40 even though I tried to call the server.)

 

Any help is appreciated.

 

 

<VoIP_Registrations>

<VoIP_Registration>

<Display>IVR</Display>

<Protocol>SIP</Protocol>

<RegServer>peer3.thevoicemanager.com</RegServer>

<RegClient>9787163929@216.86.41.167</RegClient>

<LocalAlias>9787163929@216.86.41.167</LocalAlias>

<Expires>1800</Expires>

</VoIP_Registration>

</VoIP_Registrations>

<VoIP_Authentications>

<VoIP_Authentication>

<Display>Fusion</Display>

<Realm></Realm>

<Identity></Identity>

<AuthUsername></AuthUsername>

<AuthPassword></AuthPassword>

</VoIP_Authentication>

</VoIP_Authentications>

VG_Fusion_CallLog.zip

[SupportTeam]

You need to speak to SIP provider and to the people who are responsible for network connectivity.

 

You say that the provided SIP trunk is "static IP with no authentication" - which is the best approach, but when using this approach you need to make sure the network routing between your system and the SIP service provider will always correctly forward their SIP messages. The static routes to achieve that need to be put in place.

 

But do you think that maybe the network connectivity had not been set up using static routing, and you require that REGISTER messages are re-sent to the SIP provider on periodic basis?

This recurring messaging would be needed if the static routing to forward SIP messages from SIP provider to your system has not been set up, and the NATs in network path need their routing tables periodically refreshed.

If this is the case then the SIP provider should either set up your SIP trunk so that this REGISTER is successful without Authentication, or provide you with user/password to use to perform Register with Authentication.

Right now the REGISTER request is replied to with a 401-Unauthorized, and VoiceGuide does not reply to that with another REGISTER as no Authentication user/password was set.

And further Register attempts are not made once the Register failed.

 

 

WireShark trace shows that the first call is sent by them to IVR without the REGISTER being authenticated, which indicates that this SIP trunk right now does not require Authentication to function.

But the re-sending of REGISTERs may be needed if your networking connection to SIP provided does not have the static routing set up in place to always forward their INVITES to your system.

[iTime]

Thanks for the quick reply.

You have mentioned,

If so then the SIP provider should either be set up your SIP trunk so that this REGISTER is successful without Authentication, or provide you with user/password to use when Authenticating.

 

Does this mean I can ask for username and password even for Static IP environment? The service provider mentioned authentication can be set, but with dynamic IP address.

[iTime]

I have one more question regarding your comment,

But the re-sending of REGISTERs may be needed if your networking connection to SIP provided does not have the static routing set up in place to always forward their INVITES to your system.

 

 

How would I achieve this? Can I configure VG to do so?

[SupportTeam]

First of all:

 

Your network connectivity between the SIP provider and your system needs to be setup in such a way that the INVITE messages from your SIP provider arrive at your system.

 

If you have a "static IP with no authentication" SIP trunk then this should be happening without any REGISTER messages from your system.

 

It's possible that by sending the REGISTER message to SIP provider you are enabling for the return connection from SIP provider to you to work, where as it would not work otherwise.

Right now you have VoiceGuide system set up to send out a REGISTER message on service restart.

If after a few minutes pause in communications you no longer see INVITE messages from VoIP provider then, then it looks like you do not have static IP routing in place.

If after a few minutes pause in communications you no longer see INVITE messages from VoIP provider then, then it looks like the initial sending of REGISTER message enabled the routing through the NATs in between your system and the SIP provider, and these routes timed out after a few minutes.

 

 

You need to speak to people who are responsible setting up network connectivity. Network routing needs to be set so that messages from VoIP service provider always arrive at your system.

 

Try removing the "VoIP_Registration" entry in VoiceGuide's Config.xml and restating the VoiceGuide service.

Most likely you will then not see any VoIP calls arriving at your system - as static IP routing has not been set up.

 

 

If you cannot get your network administrators to set up this static IP routing, then you can try workarounds - which can involve repeated sending of REGISTER messages from your system to the SIP provider. This repeated sending of REGISTER messages can refresh NAT tables. If the static routing has not been set up then this constant refreshing of NAT tables may be necessary to allow communications.

 

 

Does this mean I can ask for username and password even for Static IP environment?

 

It's not usually needed in 'static IP routing' setup, so it would be an unusual request to make. You can see if your SIP provider will give you this option, if they do then you can use the Register(+Authenticate is needed) as a workaround until the static route has been set up properly.

 

The 're-sending or REGISTERs' is done if you have setup the "VoIP_Registration" entry in VoiceGuide's Config.xml, and the SIP service provider is confirming the registrations are successful.

Usually for registration to be successful you need to Authenticate as well - and to do that you need to setup the "VoIP_Authentication" entry in VoiceGuide's Config.xml as well.

Right now your SIP service provider is sending a '401-Unauthorized' in response to your REGISTER message, so you need to ask your SIP provider for user/password which you can use to Authenticate (or ask them to not require user/password, and just accept REGISTER based on your IP address, but most would not allow this)