Cisco Support Community
cancel
Showing results for 
Search instead for 
Did you mean: 
Announcements

Welcome to Cisco Support Community. We would love to have your feedback.

For an introduction to the new site, click here. If you'd prefer to explore, try our test area to get started. And see here for current known issues.

New Member

Spa3102 would not forward a voip call to pstn line

Good morning.
I've done the implementation provided here http://community.linksys.com/t5/VoIP-Adapters/SPA-3102-and-softphone-to-

make-calls-via-pstn-line/td-p/326390 .
It is a way to use for outgoing calls a given pstn line from anywhere I have internet (voip to pstn).
The spa3102 is connected to a router (with an active DHCP server and ip 192.168.1.1) from where it takes the internal

ip (192.168.1.3).On the same network is also a computer , connected to the router ( with ip 192.168.1.2). The spa3102

is set to bridge mode and thus inactivates the function of the router (on SPA3102), and it functions as a  simple

network device . I have  done port forwarding (from the router) to 192.168.1.3 (SPA3102) for the port 5061 (PSTN

LINE) ( but for 5060 for the LINE 1 also). I want to make calls from a voip softphone (x-lite 4) to the SPA 3102 and

this to forward the voip calls to PSTN line to which it is connected. In x-lite the SPA3102 is set as a proxy so that

i can type the phone number I want to call without being followed by the SPA3102's ip each time ( eg on  x-lite I

give call number 2101111111 instead of 2101111111 @ wanip: 5061 where wanip is the external ip of the router).
When x-lite is running on the computer that is on the same network with the SPA3102 everything works as expected. A

voip call is made from x-lite ( using as a proxy the wanip everytime, or even for test purposes the dyndns domain

that i set up for this reason), this call is passese PSTN line and the phone of the called party rings . At x-lite

COMES indication "call established ".

The problem occurs when I do the same procedure from x-lite installed on a computer belonging to another network (

e.g. in another building with its own internet connection , own router, own computer , etc. ) . Always using the

wanip the x-lite makes the voip call to the SPA3102, writes "call established" ( meaning it connected to SPA3102) but

never routed the call to the called party ( the SPA3102 did not forward voip calls it receives to the PSTN line ) .
Trying to find what 's wrong I've tried to disable all firewalls (soft and hard from all involved machines ) . The

behavior is the same either the computer that makes the successful calls is connected to the network directly to the

router  or through the port "ethernet" on the SPA3102.

What is the difference in these two voip calls to the SPA3102 and the one  " triggers "  it to forward the call to

PSTN line and the other does not ?

Thanks now for any ideas you give .

2 ACCEPTED SOLUTIONS

Accepted Solutions

Re: Spa3102 would not forward a voip call to pstn line

What is the difference in these two voip calls to the SPA3102 and the one  " triggers "  it to forward the call to

PSTN line and the other does not ?

The "difference" more than likely is the addressing used in the call sip signalling.  On the SPA3102 on the PSTN Line Tab do you have NAT Mapping Enable: Yes or No?  If NAT Mapping Enable is set to No the SPA3102 will insert a local network address in various places in the signalling.  This is for calls over the local network.  If NAT Mapping Enable is set to YES it will insert an external ip address if it knows the external ip address. 

If the SPA3102 has a static external ip address you can make that setting on the Sip Tab.  It is also possible to setup a STUN server to allow the SPA3102 to discover its external ip address.  You can use any STUN server since it is a standardized function.

Calls do not usually work correctly if the call is routed downstream from the SPA3102 thru the "ethernet" port of the SPA3102 router.  Calls generally need to be routed thru the "Internet" (WAN) port of the SPA3102.  I do not believe you are cabled thru the "ethernet" port, I am just mentioning this for anyone reviewing the posting.

Re: Spa3102 would not forward a voip call to pstn line

The audio sound problem is more than likely also associated with the overall addressing problem initially encountered.  As you may know, using the sip protocol the sip signalling exchanges ip addresses to be used for both the sip signalling and the exchange of rtp sound packets.  In addition there is an exchange of port numbers to be used for the exchange of rtp sound packets.  The sound is exchanged by two separate streams of packets, one stream in each direction.  The result is an ip address and port number for the rtp packets flowing from the SPA3102 to the softphone and a different ip address and port number for the rtp packets flowing from the softphone to the SPA3102.

In your previous posting you mentioned that you "set the minimum  EXTernal rtp port at the sip tab".  Changing the "EXT RTP Port Min:" is an unusual change to make and in my opinion would only be made in special circumstances. Actually, I ran some tests and I'm not sure exactly what that setting does.  In my tests it didn't appear to affect the rtp port number used in a predictable manner.

The common changes to make for audio problems typically would be to setup a STUN server.  A STUN server is an external server that echos back to the initial sender the external ip address and port number that the STUN server received with the message received by the server.  This allows the sender (SPA3102 or softphone) to determine its external ip address and external port numbers for both the sip signalling and rtp packets.

A STUN server is commonly recommended to be setup with the following settings in the SPA3102:

PSTN Line Tab:

NAT Mapping Enable: Yes

Sip Tab:

Handle VIA received: yes

Handle VIA rport: yes

Insert VIA received: yes

Insert VIA rport: yes

Substitute VIA Addr: yes

Send Resp To Src Port: yes

STUN Enable: yes

STUN Server:

The following web page has a list of "Public STUN Servers"

http://www.voip-info.org/wiki/view/STUN

You are using CounterPath's XLite softphone.  stun.counterpath.net  is a STUN server on the list.

I see XLite also has a setting to use a STUN server on the "Topology" tab.

5 REPLIES

Re: Spa3102 would not forward a voip call to pstn line

What is the difference in these two voip calls to the SPA3102 and the one  " triggers "  it to forward the call to

PSTN line and the other does not ?

The "difference" more than likely is the addressing used in the call sip signalling.  On the SPA3102 on the PSTN Line Tab do you have NAT Mapping Enable: Yes or No?  If NAT Mapping Enable is set to No the SPA3102 will insert a local network address in various places in the signalling.  This is for calls over the local network.  If NAT Mapping Enable is set to YES it will insert an external ip address if it knows the external ip address. 

If the SPA3102 has a static external ip address you can make that setting on the Sip Tab.  It is also possible to setup a STUN server to allow the SPA3102 to discover its external ip address.  You can use any STUN server since it is a standardized function.

Calls do not usually work correctly if the call is routed downstream from the SPA3102 thru the "ethernet" port of the SPA3102 router.  Calls generally need to be routed thru the "Internet" (WAN) port of the SPA3102.  I do not believe you are cabled thru the "ethernet" port, I am just mentioning this for anyone reviewing the posting.

New Member

Re: Spa3102 would not forward a voip call to pstn line

You were right. Your answer was very helpful. The only thing  that i had to do beyond your  advice , was to set the minimum  EXTernal rtp port at the sip tab.

Thanks again

        P.S. the SPA3102 was connected through the internet port all the time 

New Member

Re: Spa3102 would not forward a voip call to pstn line

I am afraid that an other issue arose. After the above hint the connection from the softphone to the called party is established (through the SPA3102) , however there is no sound to any of the two ends of the call connection .

Again, when the test is performed from a sofphone on a pc on the same internal network (always using the external ip on the softphone though) sound travels both ways without any problems.

So i am asking again for any ideas about this new issue.

Re: Spa3102 would not forward a voip call to pstn line

The audio sound problem is more than likely also associated with the overall addressing problem initially encountered.  As you may know, using the sip protocol the sip signalling exchanges ip addresses to be used for both the sip signalling and the exchange of rtp sound packets.  In addition there is an exchange of port numbers to be used for the exchange of rtp sound packets.  The sound is exchanged by two separate streams of packets, one stream in each direction.  The result is an ip address and port number for the rtp packets flowing from the SPA3102 to the softphone and a different ip address and port number for the rtp packets flowing from the softphone to the SPA3102.

In your previous posting you mentioned that you "set the minimum  EXTernal rtp port at the sip tab".  Changing the "EXT RTP Port Min:" is an unusual change to make and in my opinion would only be made in special circumstances. Actually, I ran some tests and I'm not sure exactly what that setting does.  In my tests it didn't appear to affect the rtp port number used in a predictable manner.

The common changes to make for audio problems typically would be to setup a STUN server.  A STUN server is an external server that echos back to the initial sender the external ip address and port number that the STUN server received with the message received by the server.  This allows the sender (SPA3102 or softphone) to determine its external ip address and external port numbers for both the sip signalling and rtp packets.

A STUN server is commonly recommended to be setup with the following settings in the SPA3102:

PSTN Line Tab:

NAT Mapping Enable: Yes

Sip Tab:

Handle VIA received: yes

Handle VIA rport: yes

Insert VIA received: yes

Insert VIA rport: yes

Substitute VIA Addr: yes

Send Resp To Src Port: yes

STUN Enable: yes

STUN Server:

The following web page has a list of "Public STUN Servers"

http://www.voip-info.org/wiki/view/STUN

You are using CounterPath's XLite softphone.  stun.counterpath.net  is a STUN server on the list.

I see XLite also has a setting to use a STUN server on the "Topology" tab.

New Member

Re: Spa3102 would not forward a voip call to pstn line

Once again you were right. My mistake was that although i had set up the stun server for the SPA3102 i hadn't done the same for the softphone (x-lite).

That was the only change i had to do for the whole setup to run smoothly.

Thanks again, your help is highly appreciated.

2024
Views
0
Helpful
5
Replies