Hi, Andrew!

What WebRTC client are you using? Could you capture the SIP messages exchanged between the two endpoints?

Best regards,

Răzvan Crainea
OpenSIPS Core Developer / SIPhub CTO
http://www.opensips-solutions.com / https://www.siphub.com

On 11/29/23 17:37, Andrew Colin via Users wrote:
Correct I am using WSS

I have tested with SIP as well and had no issues

*From: *Bogdan-Andrei Iancu <bog...@opensips.org>
*Date: *Wednesday, 29 November 2023 at 15:33
*To: *Andrew Colin <andrew.co...@ipcortex.co.uk>, users@lists.opensips.org <users@lists.opensips.org>
*Subject: *Re: [OpenSIPS-Users] Strange Nat issue

The routing of the ACK is done accordingly to the routing info in the ACK itself (like RURI and Route hdrs). To see which is the next hop (as SIP for the ACK), after the successful loose_route(), log the $ru and $du... And I understand you are actually using WSS, right ?

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

   https://www.opensips-solutions.com  <https://www.opensips-solutions.com>

   https://www.siphub.com  <https://www.siphub.com>

On 29.11.2023 17:27, Andrew Colin wrote:

    Hi Bogdan,

    Seems to be in the context of the ACK yes.

    Why would I be seeing proto 5 if we are using WSS then?

    Kind Regards

    *From: *Bogdan-Andrei Iancu <bog...@opensips.org>
    <mailto:bog...@opensips.org>
    *Date: *Wednesday, 29 November 2023 at 15:17
    *To: *users@lists.opensips.org <mailto:users@lists.opensips.org>
    <users@lists.opensips.org> <mailto:users@lists.opensips.org>, Andrew
    Colin <andrew.co...@ipcortex.co.uk> <mailto:andrew.co...@ipcortex.co.uk>
    *Subject: *Re: [OpenSIPS-Users] Strange Nat issue

    Hi Andrew,

    Proto 5 is WS (not WSS). Can you confirm if the error occurs in the
    context of the ACK ?

    Regards,


    Bogdan-Andrei Iancu

    OpenSIPS Founder and Developer

       https://www.opensips-solutions.com  <https://www.opensips-solutions.com>

       https://www.siphub.com  <https://www.siphub.com>

    On 29.11.2023 15:15, Andrew Colin via Users wrote:

        Hi All,

        Recently deployed opensips into AWS and when we make calls
        between 2 webrtc clients I keep seeing this error in the logs
        and the call eventually drops after 32 seconds

        ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 5  (no
        corresponding listening socket)

        ERROR:tm:t_forward_nonack: failure to add branches

        Normal SIP to SIP calls do not have the issue




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