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chan_pjsip: also return all codecs on empty re-INVITE for late offers 

We should also return all codecs on an re-INVITE without SDP for a
call that used late offer (e.g. no SDP in the initial INVITE, SDP
in the ACK). Bugfix for feature introduced in ASTERISK-30193
(https://issues.asterisk.org/jira/browse/ASTERISK-30193)

Migration from previous gerrit change that was not merged.
This commit is contained in:
Henning Westerholt 2023-05-03 09:19:35 +00:00
parent 8d3d8fcdcd
commit 37a41a2a20
1 changed files with 5 additions and 3 deletions
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8
res/res_pjsip_session.c
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@ -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;
}
}