Hi Tim, All aspects of SDP negotiation are between Asterisk and the Client, on that last call leg (the "B"-leg). Asterisk states to the client how it will handle DTMF and chooses a payload type (SDP offer), and the client responds to Asterisk in kind (SDP answer). If any adaptation is required between the "B" and "A"-leg, that is up to Asterisk to perform, so it must stay in the path.
What you seem to desire is a fully transparent B2BUA mode from Asterisk where it has no stake in the media parameters whatsoever, and simply passes it along what it receives from the counterparties on both the "A" and "B" legs. As far as I know, this does not exist. -- Alex > On Jul 16, 2024, at 3:54 AM, Tim Bowyer via sr-users > <[email protected]> wrote: > > Hello Everyone, > I’ve been grappling with this issue for some months, time to raise the white > flag. > Essentially I need to work out a way to configure Kamailio/RTPengine to > normalise telephone-events to 101 earlier in the piece. > > • Call comes in from Carrier with INVITE w/SDP, telephone-event payload > of 96. > • INVITE sent to Asterisk with the same SDP, payload = 96 > • INVITE sent from Asterisk to Client with payload = 101 > • 200 OK w/SDP from Client has payload = 101 > • 200 OK w/SDP to Kamailio has payload = 96 > • 200 OK w/SDP to Carrier has payload = 96 > • Asterisk (with Direct Media enabled) sends INVITE to Kamailio with > payload = 96, and new media IP of Client > • Kamailio passes this onto Carrier with media IP of RTPengine > • 200 OK w/SDP from Carrier has payload = 96 sent to Kamailio > • Kamailio passes this 200 OK to Asterisk, which responds with ACK and > doesn’t do anything further > +---------+ +----------+ +---------+ > +---------+ > | Carrier | <----1-----> | Kamailio | <-----2----> | Asterisk| <----3------>| > Client | > | | | | | | | > | > +---------+ +----------+ +---------+ > +---------+ > ^ | | > | > | | |3| > | > | | | > | > +------5---------> +-----------+ > <-----------------+------------4-----------+ > | RTPengine | > +-----------+ > Big issue is, the Client is expecting telephone-events with a payload of > 101, whereas RTPengine is passing it 96 from the Carrier. > > • Inbound DTMF only works at client-side if Asterisk stays in path (far > from ideal). > • The issue is exacerbated when calls are sent back out to the same or > another carrier (forward scenario). > • There’s nothing I can seem to do in Asterisk to change how it handles > DTMF in a direct media/re-invite setting. > Any ideas? > Cheers, > Tim Bowyer > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to [email protected] > Important: keep the mailing list in the recipients, do not reply only to the > sender! > Edit mailing list options or unsubscribe: -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800 __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to [email protected] Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
