<p>Henning Westerholt has uploaded this change for <strong>review</strong>.</p><p><a href="https://gerrit.asterisk.org/c/asterisk/+/19979">View Change</a></p><pre style="font-family: monospace,monospace; white-space: pre-wrap;">chan_pjsip:<br><br>Bugfix for feature introduced in ASTERISK-30193<br>We should also return all codecs on an re-INVITE without SDP<br>for a call that used late offer (e.g. no SDP in the initial INVITE,<br>SDP in the ACK).<br><br>ASTERISK-30473<br><br>Change-Id: I0c132202eef4f6746212559d0e936951e17adc6c<br>---<br>M res/res_pjsip_session.c<br>1 file changed, 21 insertions(+), 3 deletions(-)<br><br></pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/79/19979/1</pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;"><span>diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c</span><br><span>index ae2e0e4..63177e1 100644</span><br><span>--- a/res/res_pjsip_session.c</span><br><span>+++ b/res/res_pjsip_session.c</span><br><span>@@ -5348,14 +5348,16 @@</span><br><span>   /* Some devices send a re-INVITE offer with empty SDP. Asterisk by default return</span><br><span>     * an answer with the current used codecs, which is not strictly compliant to RFC</span><br><span>     * 3261 (SHOULD requirement). So we detect this condition and include all</span><br><span style="color: hsl(0, 100%, 40%);">-        * configured codecs in the answer if the workaround is activated.</span><br><span style="color: hsl(120, 100%, 40%);">+     * configured codecs in the answer if the workaround is activated. The actual</span><br><span style="color: hsl(120, 100%, 40%);">+  * logic is in the create_local_sdp function. We can't detect here that we have</span><br><span style="color: hsl(120, 100%, 40%);">+    * no SDP body in the INVITE, as we don't have access to the message.</span><br><span>     */</span><br><span>  if (inv->invite_tsx && inv->state == PJSIP_INV_STATE_CONFIRMED</span><br><span>                         && inv->invite_tsx->method.id == PJSIP_INVITE_METHOD) {</span><br><span>                ast_trace(-1, "re-INVITE\n");</span><br><span style="color: hsl(0, 100%, 40%);">-         if (inv->invite_tsx->role == PJSIP_ROLE_UAS && !pjmedia_sdp_neg_was_answer_remote(inv->neg)</span><br><span style="color: hsl(120, 100%, 40%);">+          if (inv->invite_tsx->role == PJSIP_ROLE_UAS</span><br><span>                            && ast_sip_get_all_codecs_on_empty_reinvite()) {</span><br><span style="color: hsl(0, 100%, 40%);">-                        ast_trace(-1, "no codecs in re-INIVTE, include all codecs in the answer\n");</span><br><span style="color: hsl(120, 100%, 40%);">+                        ast_trace(-1, "UAS role, include all codecs in the answer on empty SDP\n");</span><br><span>                        ignore_active_stream_topology = 1;</span><br><span>           }</span><br><span>    }</span><br><span></span><br></pre><p>To view, visit <a href="https://gerrit.asterisk.org/c/asterisk/+/19979">change 19979</a>. To unsubscribe, or for help writing mail filters, visit <a href="https://gerrit.asterisk.org/settings">settings</a>.</p><div itemscope itemtype="http://schema.org/EmailMessage"><div itemscope itemprop="action" itemtype="http://schema.org/ViewAction"><link itemprop="url" href="https://gerrit.asterisk.org/c/asterisk/+/19979"/><meta itemprop="name" content="View Change"/></div></div>

<div style="display:none"> Gerrit-Project: asterisk </div>
<div style="display:none"> Gerrit-Branch: 18 </div>
<div style="display:none"> Gerrit-Change-Id: I0c132202eef4f6746212559d0e936951e17adc6c </div>
<div style="display:none"> Gerrit-Change-Number: 19979 </div>
<div style="display:none"> Gerrit-PatchSet: 1 </div>
<div style="display:none"> Gerrit-Owner: Henning Westerholt <hw@gilawa.com> </div>
<div style="display:none"> Gerrit-MessageType: newchange </div>