Is there a FAQ file? 'Cause I have lots of Qs

Discussion in 'VOIP' started by Joe, Aug 3, 2005.

  1. Joe

    Joe Guest

    Hi,

    I have the most basic questions. I also have a Sipura SP841, an account
    on Broadvoice, and no phone service. ;) I'd like to trade in the
    questions for some answers/knowledge and maybe even, as a bonus, a
    working VOIP system.

    Basically, the 841 is here behind my Linux NAT/firewall (netfilter) host
    and my net connection goes through Comcast cable. The phone has a fixed
    192.168 address and the firewall uses DHCP with Comcast. I can see the
    phone's menu from browsers on other machines on my network, but I can't
    call out and incoming calls go straight to voice mail with the "Joe is
    busy" message.

    Netfilter is set to ACCEPT udp from the external interface on ports 69,
    5060:5063, and 10000:20000.

    Is there a FAQ for this newsgroup? A good source for the very rudiments
    of application? I don't mind starting from the start to learn this
    stuff. Obviously jumping right in wasn't the optimal approach.

    TIA,

    Joe
    Joe, Aug 3, 2005
    #1
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  2. Joe:

    Have you tryed starting with the FWD?

    http://www.freeworlddialup.com

    They have excellent help for beginners, and they have Sipura-specific
    configurations.

    -Ramon
    Ramon F Herrera, Aug 3, 2005
    #2
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  3. Joe wrote:
    > Hi,
    >
    > I have the most basic questions. I also have a Sipura SP841, an account
    > on Broadvoice, and no phone service. ;) I'd like to trade in the
    > questions for some answers/knowledge and maybe even, as a bonus, a
    > working VOIP system.
    >
    > Basically, the 841 is here behind my Linux NAT/firewall (netfilter) host
    > and my net connection goes through Comcast cable. The phone has a fixed
    > 192.168 address and the firewall uses DHCP with Comcast. I can see the
    > phone's menu from browsers on other machines on my network, but I can't
    > call out and incoming calls go straight to voice mail with the "Joe is
    > busy" message.


    Welcome to the "SIP+NAT Nightmares Club" :)

    > Netfilter is set to ACCEPT udp from the external interface on ports 69,
    > 5060:5063, and 10000:20000.
    >
    > Is there a FAQ for this newsgroup? A good source for the very rudiments
    > of application? I don't mind starting from the start to learn this
    > stuff. Obviously jumping right in wasn't the optimal approach.


    OK, here's the deal. The SIP protocol is very NAT-unfriendly, for
    basically two reasons:

    1. As it name says, it only deals with session initiation; the actual
    voice packets are transported with a separate protocol called RTP,
    which uses UDP but with different port numbers. Even when the SIP
    issues are ironed out, sometimes one can't hear audio due to RTP
    problems.

    2. SIP messages contain, ASCII-encoded, several parameters that are
    affected by NAT: IP addresses and port numbers (both for SIP and RTP).
    >From how you describe your case, the connection already fails at SIP

    level.

    Some relief comes from enhancements made to the protocol and additional
    helper protocols, such as STUN (http://www.ietf.org/rfc/rfc3489.txt),
    that allows a NATted client to find out the external address of its
    router (so that it may communicate to the peer THAT address, and not
    its own on the LAN) and the type of NAT (to attempt educated guesses on
    port mappings). Still, it's a struggle. To some extent, this is a
    repeat of the headaches with FTP behind NAT, the attempts at fixing
    them with "Passive FTP" etc. Only, worse.

    I'm not familiar with the SPA-841 but I have used SPA-2100 and
    SPA-3000, and I think most settings are similar. In my SIP settings,
    under "NAT Support Parameters" I have set all parameters to "Yes"
    except "Send Resp To Src Port:" which is "No". As STUN server I use
    stun.fwdnet.net:3478 . "NAT Keep Alive Intvl:" is set to 15.

    Then, in the "Line 1" and "PSTN" screens, under "NAT Settings" I have
    set to "Yes" both "NAT Mapping Enable:" and "NAT Keep Alive Enable:";
    the "NAT Keep Alive Msg:" is better left empty rather than the default
    "$NOTIFY". And I recommend you also to set "Symmetric RTP:" to "Yes",
    otherwise you might lose audio in one direction, or both.

    Also, for anything related to Sipura check out the excellent site
    http://voxilla.com/ and its forums, which are moderated by competent
    people.

    Good luck,

    Enzo
    Enzo Michelangeli, Aug 3, 2005
    #3
  4. Joe

    Joe Guest

    Enzo,

    Thanks for taking time to help me.

    Do you know if there are specific settings I need to use for the NAT
    machine? The only change I introduced for the phone was the ACCEPT udp
    on the 69, 5060:5063 and 10000:20000 ports.

    > I'm not familiar with the SPA-841 but I have used SPA-2100 and
    > SPA-3000, and I think most settings are similar. In my SIP settings,
    > under "NAT Support Parameters" I have set all parameters to "Yes"
    > except "Send Resp To Src Port:" which is "No". As STUN server I use
    > stun.fwdnet.net:3478 . "NAT Keep Alive Intvl:" is set to 15.
    >
    > Then, in the "Line 1" and "PSTN" screens, under "NAT Settings" I have
    > set to "Yes" both "NAT Mapping Enable:" and "NAT Keep Alive Enable:";
    > the "NAT Keep Alive Msg:" is better left empty rather than the default
    > "$NOTIFY". And I recommend you also to set "Symmetric RTP:" to "Yes",
    > otherwise you might lose audio in one direction, or both.


    I've made these changes as I see them on the SPA-841 screens. I've
    progressed from "Not Registered" and no lights on the unit to
    "Registration Failed" and all the lights on the unit glowing amber.
    Considering where I started, I call that progress!

    > Also, for anything related to Sipura check out the excellent site
    > http://voxilla.com/ and its forums, which are moderated by competent
    > people.


    I also found some settings from the config tool, but there is some
    trouble because when I try to upload those settings I get an error on
    the config page in the browser.

    I'd be better at debugging if I had some idea how this is supposed to
    work. When I debug a web app, I understand basics like http, rmi, etc.
    I'm poking at this like a voodoo doll, trying to get the right
    combination of pins to effect my desired result. Not exactly an optimal
    process, eh? ;)

    Thanks,

    Joe
    Joe, Aug 5, 2005
    #4
  5. Joe wrote:
    > Enzo,
    >
    > Thanks for taking time to help me.
    >
    > Do you know if there are specific settings I need to use for the NAT
    > machine? The only change I introduced for the phone was the ACCEPT udp
    > on the 69, 5060:5063 and 10000:20000 ports.


    This should be OK depending on the SPA-841's settings. Usually each
    subunit (e.g., each FXS port in a SPA-2100) has its own SIP port, and
    the SIP ports start from the default 5060, so you should be in good
    shape for that. For the RTP ports, the range is specified somewhere in
    the Sipura setup web screens. For example, in the SPA-3000 there are
    two fields in the "SIP" screen labelled "RTP Port Min:" and "RTP Port
    Max:" (in my case, I set them to 16384 and 16482 respectively, which
    falls inside the range you gave to tour router).
    But anyway RTP gets in the picture at a later stage, after the session
    initiation, so this is not what makes your registration fail.

    > I've made these changes as I see them on the SPA-841 screens. I've
    > progressed from "Not Registered" and no lights on the unit to
    > "Registration Failed" and all the lights on the unit glowing amber.
    > Considering where I started, I call that progress!


    Are you sure your username and password are correct? Why don't you test
    the account first with a softphone, such as Xten X-Lite or the simpler
    Firefly (https://www.virbiage.com/download.php , under "Download the
    latest third-party version of the Virbiage Softphone here."

    > > Also, for anything related to Sipura check out the excellent site
    > > http://voxilla.com/ and its forums, which are moderated by competent
    > > people.

    >
    > I also found some settings from the config tool, but there is some
    > trouble because when I try to upload those settings I get an error on
    > the config page in the browser.


    Which error, exactly? And have you used a tool specific to the SPA-841?
    As far as I know, tools that work for SPA-2000 / SPA-2100 probably
    won't work for the SPA-841.

    > I'd be better at debugging if I had some idea how this is supposed to
    > work. When I debug a web app, I understand basics like http, rmi, etc.
    > I'm poking at this like a voodoo doll, trying to get the right
    > combination of pins to effect my desired result. Not exactly an optimal
    > process, eh? ;)


    If you want a quick tutorial about SIP, you may try this one, which
    helped me a lot when I began to deal with SIP):

    http://www.iptel.org/ser/doc/sip_intro/sip_introduction.html

    It should at least give you a rough idea of the "big picture". Then you
    might get Ethereal (www.ethereal.com) and use it to sniff the UDP
    packets to understand why the "REGISTER" transactions fail. The biggest
    problem is that the PC running Ethereal (placing the LAN card in
    "promiscuous mode") should be able to "see" the packets between SPA-841
    and SIP proxy, but in these days LANs are almost all switched... You
    should find an old non-switching hub, or configure the PC as a router
    and put it in the path between the SPA-841 and the Internet, etc.

    Cheers --

    Enzo
    Enzo Michelangeli, Aug 6, 2005
    #5
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