[OpenSIPS-Users] TCP to UDP proxy
Simon Quigley
squigley at versature.com
Wed Nov 6 20:15:04 CET 2013
Hi,
I've done some more testing, and it seems the issue I'm having with the
NATed contact is related to the UA I'm using.
If I use empathy/telepathy-sofiasip, then the contact isn't being mangled
properly, which causes it to send the 200 OK packet after answering,
receive no ACK, and continue resending the 200 OK, until it gives up and
sends a BYE.
The same behaviour occurs if I use sipdroid, however it doesn't send a BYE
when it doesn't receive an ACK, but the call still isn't up properly, and
when I do hang up, it doesn't send a BYE, so the originating UA (and
switch) thinks the call is still up.
However, if I use the built in Android SIP client, it's contact is mangled
properly, and so the OK / ACK process works properly.
How can I debug to see what the fix nated contact function is doing, to see
if I can work out what the differences between these UA's packets are?
Thanks
On Fri, Nov 1, 2013 at 5:44 PM, Simon Quigley <squigley at versature.com>wrote:
> Hi Bogdan,
>
> Thanks for your response, your suggestion was much appreciated, and gave
> me a good direction to work in.
>
> You are correct, I was trying to mix stateless and stateful routing
> together, I realised after watching the webinar.
>
> I've made some changes to my configuration file to use record_route() and
> loose_route(), in a few different ways, but it hasn't made any difference.
>
> The 200 OK is always sent back to the switch containing the non routable
> IP in the contact of the device, rather than fix_nated_contact changing it
> to the IP the packet arrives from, and it's also failing to match it and
> translate it for the ACK response to get to the device.
>
> I've enabled debugging, and looking through the output, I think it's an
> issue with matching the dialogue.
>
> Here's the config I'm using now:
>
>
> listen=udp:[internal VM IP]:5060
> listen=tcp:[internal VM IP]:5060
>
> disable_tcp=no
>
> advertised_address="[dns name]"
> alias="[dns name]"
>
>
> route{
> if (src_ip != "[switch ip]"){
> # from device
> if (!is_method("REGISTER|MESSAGE")) {
>
> if (is_method("INVITE")) {
> fix_nated_sdp("7");
> record_route();
>
> }
> else if (is_method("OPTIONS")){
> exit;
> }
> else {
> loose_route();
> fix_nated_contact();
> if (has_totag()){
> fix_nated_sdp("7");
> }
> t_relay("udp:[switch ip]:5060");
>
> }
> }
> }
>
> else if (src_ip == "[switch ip]"){
> # from switch
> if (is_method("INVITE")) {
> fix_nated_contact();
> fix_nated_sdp("7");
> record_route();
> }
> else {
> loose_route();
>
> }
>
> # mangle to TCP
> $du = $ru + ";transport=tcp";
>
> t_relay();
> }
> exit;
> }
>
> Is there something obvious that I'm doing wrong?
>
> Thanks again for your help.
>
>
> On Wed, Oct 30, 2013 at 4:52 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>wrote:
>
>> Hello Simon,
>>
>> Most probably it is a mistake on the way you do record_routing (for
>> initial INVITE) and loose_route (for sequential requests). All sequential
>> requests are driven by this mechanism. If the case, check how it suppose to
>> work :
>> http://www.opensips.org/Documentation/Webinars#toc12 (5.5)
>>
>> Basically you need to do record_route() for INVITE and loose_route() for
>> ACK - OpenSIPS will take care of changing the interface for the ACK after
>> the loose_route() call.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>>
>> On 10/25/2013 07:42 PM, Simon Quigley wrote:
>>
>> Hello,
>>
>> I'm trying to use OpenSIPS as a TCP to UDP proxy, (with the registrar
>> module disabled) connecting all sorts of devices from different IP
>> addresses over TCP to a UDP switch.
>>
>> So far I've managed to get OpenSIPS configured to change the contact
>> header, and the SDP, append the transport, and forward the packets
>> correctly, for making an outbound call (from TCP device to UDP switch).
>>
>> I've got incoming to work somewhat (from UDP switch to TCP device), the
>> INVITE will pass, and the TRYING/RINGING gets back to the switch, however
>> the issue I'm having is that the OK packet from the TCP device is sent to
>> the switch, which responds with the ACK, however the contact header is
>> wrong, and OpenSIPS is trying to sent the ACK to the internal non routable
>> IP of the device.
>>
>> As a result the call comes up, but the device has an ACK timeout and
>> drops the call.
>>
>> I'm having difficulties working out how to get the contact header
>> mangled sufficiently for the OK/ACK to contain the routable IP of the TCP
>> device, rather than it's internal/behind NAT IP.
>>
>> I've tried removing the header, using loose_route, using t_relay, and
>> (am currently attempting) to use route(relay).
>>
>> Also of relevance, is the fact that I'm running OpenSIPS on a Google
>> Compute VM, so it's behind 1 to 1 NAT, with a non routable IP on the VM,
>> but I have assigned a static routable IP to the 1 to 1 NAT, and I have DNS
>> pointing to this routable IP.
>>
>> Here's my config file, excluding default values/routes etc:
>>
>> listen=udp:[internal VM IP]:5060
>> listen=tcp:[internal VM IP]:5060
>>
>> disable_tcp=no
>>
>> advertised_address="[DNS name]"
>> alias="[DNS name]"
>>
>> route{
>> if (src_ip != "[switch ip]"){
>> # from device
>> if (is_method("INVITE")) {
>> fix_nated_sdp("7");
>> }
>> else if (is_method("OPTIONS")){
>> exit;
>> }
>>
>> fix_nated_contact();
>> forward ("udp:[switch ip]:5060");
>> }
>>
>> else if (src_ip == "[switch ip]"){
>> # from switch
>> if (is_method("INVITE")) {
>> fix_nated_sdp("7");
>> }
>> else if (is_method("ACK")) {
>> fix_nated_contact();
>> }
>>
>> # mangle to TCP
>> $du = $ru + ";transport=tcp";
>>
>> route(relay);
>> }
>>
>> exit;
>> }
>>
>> Is the issue perhaps related to the fact I'm not recording a route,
>> so it doesn't know how to correctly route the response, or am I just
>> missing something obvious?
>>
>> Thanks,
>> Simon
>>
>> --
>> Simon Quigley, Systems Engineer
>> Versature Corp. | 877-498-3772 x128
>>
>> Follow us on Twitter - http://twitter.com/Versature
>> Check out the Versature Blog - http://inside.versature.com
>>
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> Simon Quigley, Systems Engineer
> Versature Corp. | 877-498-3772 x128
>
> Follow us on Twitter - http://twitter.com/Versature
> Check out the Versature Blog - http://inside.versature.com
>
--
Simon Quigley, Systems Engineer
Versature Corp. | 877-498-3772 x128
Follow us on Twitter - http://twitter.com/Versature
Check out the Versature Blog - http://inside.versature.com
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