RV082 VoIP/SIP behind NAT

Discussion in 'Cisco Small Business Routers and VPN Solutions' started by jsturtevant, Nov 26, 2006.

  1. jsturtevant

    jsturtevant LI Guru Member

    I am trying to have a group of Polycom SIP phones connect to a remote SIP server. These phones have been working fine behind a WRT54g. The only thing that has changed is to replace the WRT with the RV082.

    When the phones send the outbound SIP message from the phone which leaves the phone as UDP:5060 it is sent to the server as 55001. The server returns the reply on 55001.

    With the WRT the same message stays on 5060 all the way through.

    Why is the RV082 changing the port number?

    Any thoughts?
  2. tdnyrfan

    tdnyrfan LI Guru Member

    It may have something to do with Statefull Packet Inspection, if you can turn that off see if that helps. I'm having the same thing happen with my Viatalk PAP2 and they said that spi needs to be turned off.
  3. ed001

    ed001 Network Guru Member

    I cant get to my 82's right now but I think binding a SIP to a single interface may solve the problem. I've done this with a client who uses Hotbricks and it seemed to solve a similar problem. Unfortunately your VOIP won't be Load balanced.
  4. jsturtevant

    jsturtevant LI Guru Member

    I'm only using one WAN interface for now.

    any insight you might have would be appreciated..
  5. eric_stewart

    eric_stewart Super Moderator Staff Member Member

    That's interesting. I think you've just described (maybe) an issue that I'm having. I observed that my VoIP SIP adapter doesn't function behind my RV042, but is fine on the uplink between the WAN1 interface and my upstream Cisco PIX501 which is acting as my edge firewall on my DSL link.

    I was wondering if maybe putting it in the DMZ (WAN2) interface might solve the issue too. If it does, this might be good workaround for you since you said in a later post on this subject that you aren't using both WAN interfaces. Let me know if you would like me to test it and I can try it tomorrow (Monday). My timezone is GMT -5 BTW.

    Here's a link to my network diagram (MS PowerPoint). The site is hosted on my Linux box which is in the noted DMZ.


  6. jsturtevant

    jsturtevant LI Guru Member

    Eric, yes, I think you may be seeing the same behavior....

    The SIP devices register and can make outbound calls, but they are unavailable for incoming calls (except during very brief windows).

    I can't use DMZ as I have 10-15 phones in an office environment.

    You might want to consider ditching the RV082 for a WRT54GL or Buffalo and the DD-WRT firmware...

    My client wanted an IPSEC VPN "supported" by a major hardware vendor. Trying to get support from Linksys on this has been difficult.
  7. egyvoip

    egyvoip LI Guru Member

    For VoIP platform rtp is very important, i had that problem for long time too but it solved like this

    *Location 1
    1- NIC1-eth0<=>mainWAN1,,,,,,,,,,,,,public(IP)
    2- NIC2-eth1<=>linksys(A)LAN,,,,,,,,,internal(IP)same subnet with linksys

    you should add this route line in your server route table
    <route add -net netmask gw eth1>

    a-Linksys wan should connect a different WAN2
    b-your servers' route table shouldn't contain
    default UG 0 0 0 eth0

    *Location 2
    all device <=> linksys(B)LAN

    using this will prevent your Linksys(A) from Unstablity caused by huge rtp traffic (over 20 calls), if you connect the server under Linksys(A) and use PortFW, just give it a while and try to ping it even from same network then you will know what i ment.

    at both side try RV042 1st if you found it unstable (it will stable) then move to RV082, while i'm facing problems with RV016 it rebooting by it self 3-5 times a day.

    Let me know the updates
  8. astyl

    astyl Guest

    SIP / NAT on RV082

    1. Configure your ATA device to request its IP Address from the RV082 DHCP Server.
    2. Enable DHCP Server on RV082, change the address range to your LAN requirements, add DNS and WINS (if you have one), ADD static entry for your ATA device with correct MAC Address.
    3. In Port Forwarding section ADD appropriate services (SIP, STUN, etc) and make the association with your ATA's IP .
    4. Leave the access rules and firewall settings default

    I have tested the above configuration in 4 different installations and it works great.

    Firmware Version:

    ADSL router (bridge mode)-RV082(WAN1(PPPoE))-RV082(LAN)-ATA FritzBox + Linksys PAP 2 (both connected to sip providers) + Siemens Voip Optipoint 410 Economy (connected to the remote site to a Siemens PBX Hipath 3750 - HG 1500).

    I have to mention that in all the above installations the RV's have also enable the VPN function
  9. cortiz50

    cortiz50 Guest

    RV082 VOIP issues

    You guys seem to describe a problem I am having but I'm not really following your explanations. I have a Vonage VoIP Rtr sitting behind the RV082 and it works intermitently. I have tried binding the port range Vonage gave me and allowing the range via the FW but still the same issue. Can you guys explan a little further for this scenario?
  10. SigAddict

    SigAddict Guest

    On your DHCP page in the RV082 set a static IP address for your PAP2's MAC address so it always gets the same address. Mine is set to Then forward the ports to that address like this:

    UDP 5060 - 5061 >
    UDP 53 - 53 >
    UDP 69 - 69 >
    UDP 10000 - 20000 >

    The PAP2 is still getting it's address via DHCP but it will always be the same address. This has worked for me rock solid for 15 months. I have my firewall and SPI enabled.
  11. jsturtevant

    jsturtevant LI Guru Member

    the issue is I have 25+ phones so I can't use port forwarding.... I "solved" the problem by dramatically increasing the frequency of registers from the Polycom phones to every 30 seconds. This approach has been used by others.

    There should be a way to configure the router to be less aggresive in disconnecting UDP sessions.

    I had the same problem with a similar Netgear router, but not with "home" routers.
  1. This site uses cookies to help personalise content, tailor your experience and to keep you logged in if you register.
    By continuing to use this site, you are consenting to our use of cookies.
    Dismiss Notice