From peter.manley@kombea.com Sat Jun 26 00:44:18 2021 From: Peter Manley To: sr-users@lists.kamailio.org Subject: [SR-Users] Guidance building a WebRTC to Asterisk Intermediate Proxy Date: Fri, 25 Jun 2021 22:44:11 +0000 Message-ID: MIME-Version: 1.0 Content-Type: multipart/mixed; boundary="===============1567703083==" --===============1567703083== Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: quoted-printable Hello, I'm working on building an RTP Proxy which will allow both regular SIP to SIP= transactions and WebRTC to SIP transactions to an Asterisk Server. SIP UAC < ---UDP/SIP --- > Kamailio/RTPEngine Proxy < ---UDP/SIP----- > Aster= isk Server AND WebRTC UAC < ---WSS--- > Kamailio/RTPEngine Proxy < ---UDP/SIP----- > Asteris= k Server The Kamailio/RTPEngine Proxy is stateless (does not use the tm or registerer = modules), and uses the Path module to replace the Route header with a Path he= ader. It works grep for SIP/UDP to SIP/UDP transactions. HOWEVER... It only partially works for WSS to SIP/UDP transactions. The REGIS= TER/401(Unauthorized), REGISTER/200(OK) transaction works well. When the Asterisk server sends a SIP Options message, I get the following err= or: Jun 25 15:47:03 VRTPENGINE kamailio[4953]: INFO: