diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c index ae2e0e4005d0d2f63726140d091851b85b202608..63177e1aac09ea783fcd3430850d3beb37258446 100644 --- a/res/res_pjsip_session.c +++ b/res/res_pjsip_session.c @@ -5348,14 +5348,16 @@ static void session_inv_on_create_offer(pjsip_inv_session *inv, pjmedia_sdp_sess /* Some devices send a re-INVITE offer with empty SDP. Asterisk by default return * an answer with the current used codecs, which is not strictly compliant to RFC * 3261 (SHOULD requirement). So we detect this condition and include all - * configured codecs in the answer if the workaround is activated. + * configured codecs in the answer if the workaround is activated. The actual + * logic is in the create_local_sdp function. We can't detect here that we have + * no SDP body in the INVITE, as we don't have access to the message. */ if (inv->invite_tsx && inv->state == PJSIP_INV_STATE_CONFIRMED && inv->invite_tsx->method.id == PJSIP_INVITE_METHOD) { ast_trace(-1, "re-INVITE\n"); - if (inv->invite_tsx->role == PJSIP_ROLE_UAS && !pjmedia_sdp_neg_was_answer_remote(inv->neg) + if (inv->invite_tsx->role == PJSIP_ROLE_UAS && ast_sip_get_all_codecs_on_empty_reinvite()) { - ast_trace(-1, "no codecs in re-INIVTE, include all codecs in the answer\n"); + ast_trace(-1, "UAS role, include all codecs in the answer on empty SDP\n"); ignore_active_stream_topology = 1; } }