Mr. Maier, thank you very much for your feedback! We provided this specifically for Telekom's "CompanyFlex" trunks which still require mediasec headers according to their website [1]. Specifically, we have to adhere to their technical specification 1TR119 [2].
> Does your patch work, too, if a server doesn't answer the Mediasec > request? We set the Require: mediasec header, so if a server does not understand this, it MUST respond with 420 Bad Extension. Nonetheless, if you have configured mediasec a server could ignore the mediasec headers and still send 2XX replies to our requests. Since the mediasec headers are static and no real security mechanism is negotiated anyways (all we need to do is satisfy Telekom's requirements), we still allow further transactions to take place (which is not how RFC 3329 intends it). However, it does affect the SDP by setting the 3ge2ae attribute, even if the server never sent us Security-Server headers. > Do you maybe plan to get asterisk ready to cope with 3 completely > independent SIP servers provided by the SRV lookup? Unfortunately, I will probably not have time to look into that in the near future. As stated above, we provided this patch to work with SIP trunks (e.g. CompanyFlex) that explicitly require mediasec. > I wasn't able to get it working. The headers you are setting > unfortunately doesn't meet the Deutsche Telekom requirements - besides > one additional bug. Thank you for testing it! We have identified similar issues (see ASTERISK-30276) and I just uploaded a patch fixing those [3]. I believe this patch fixes the issues you are seeing. In our setup, it seems to be working fine - including outgoing calls, re-registrations, and OPTIONS. Please let me know, if you are still experiencing issues with the new patch. Thanks, Max [1] https://hilfe.companyflex.de/de/grundlagen/systemvoraussetzungen [2] https://www.telekom.de/hilfe/downloads/1tr119.pdf [3] https://gerrit.asterisk.org/c/asterisk/+/19740 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev