<p>Friendly Automation <strong>submitted</strong> this change.</p><p><a href="https://gerrit.asterisk.org/c/asterisk/+/17730">View Change</a></p><div style="white-space:pre-wrap">Approvals:
Joshua Colp: Looks good to me, but someone else must approve
George Joseph: Looks good to me, approved
Friendly Automation: Approved for Submit
</div><pre style="font-family: monospace,monospace; white-space: pre-wrap;">res_pjsip_sdp_rtp: Preserve order of RTP codecs<br><br>The ast_rtp_codecs_payloads functions do not preserve the order in which<br>the payloads were specified on an incoming SDP media line. This leads to<br>a problem with the codec negotiation functionality, as the format<br>capabilities of the stream are extracted from the ast_rtp_codecs. This<br>commit moves the ast_rtp_codec to ast_format conversion to the place<br>where the order is still known.<br><br>ASTERISK-28863<br>ASTERISK-29320<br><br>Change-Id: I3aabcfed3f379c36654f59c1872c313d0cb57e25<br>---<br>M res/res_pjsip_sdp_rtp.c<br>1 file changed, 15 insertions(+), 8 deletions(-)<br><br></pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;"><span>diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c</span><br><span>index ae200a7..10db010 100644</span><br><span>--- a/res/res_pjsip_sdp_rtp.c</span><br><span>+++ b/res/res_pjsip_sdp_rtp.c</span><br><span>@@ -313,7 +313,7 @@</span><br><span> }</span><br><span> </span><br><span> static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp_media *stream, struct ast_rtp_codecs *codecs,</span><br><span style="color: hsl(0, 100%, 40%);">- struct ast_sip_session_media *session_media)</span><br><span style="color: hsl(120, 100%, 40%);">+ struct ast_sip_session_media *session_media, struct ast_format_cap *astformats)</span><br><span> {</span><br><span> pjmedia_sdp_attr *attr;</span><br><span> pjmedia_sdp_rtpmap *rtpmap;</span><br><span>@@ -329,6 +329,8 @@</span><br><span> </span><br><span> ast_rtp_codecs_payloads_initialize(codecs);</span><br><span> </span><br><span style="color: hsl(120, 100%, 40%);">+ ast_format_cap_remove_by_type(astformats, AST_MEDIA_TYPE_UNKNOWN);</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span> /* Iterate through provided formats */</span><br><span> for (i = 0; i < stream->desc.fmt_count; ++i) {</span><br><span> /* The payload is kept as a string for things like t38 but for video it is always numerical */</span><br><span>@@ -372,11 +374,19 @@</span><br><span> ast_rtp_codecs_payload_replace_format(codecs, num, format_parsed);</span><br><span> ao2_ref(format_parsed, -1);</span><br><span> }</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span> ao2_ref(format, -1);</span><br><span> }</span><br><span> }</span><br><span> }</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+ /* Parsing done, now fill the ast_format_cap struct in the correct order */</span><br><span style="color: hsl(120, 100%, 40%);">+ for (i = 0; i < stream->desc.fmt_count; ++i) {</span><br><span style="color: hsl(120, 100%, 40%);">+ if ((format = ast_rtp_codecs_get_payload_format(codecs, pj_strtoul(&stream->desc.fmt[i])))) {</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_format_cap_append(astformats, format, 0);</span><br><span style="color: hsl(120, 100%, 40%);">+ ao2_ref(format, -1);</span><br><span style="color: hsl(120, 100%, 40%);">+ }</span><br><span style="color: hsl(120, 100%, 40%);">+ }</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span> if (!tel_event && (session->dtmf == AST_SIP_DTMF_AUTO)) {</span><br><span> ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);</span><br><span> ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 0);</span><br><span>@@ -398,6 +408,7 @@</span><br><span> unsigned long framing = pj_strtoul(pj_strltrim(&attr->value));</span><br><span> if (framing && session->endpoint->media.rtp.use_ptime) {</span><br><span> ast_rtp_codecs_set_framing(codecs, framing);</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_format_cap_set_framing(astformats, framing);</span><br><span> }</span><br><span> }</span><br><span> </span><br><span>@@ -442,7 +453,6 @@</span><br><span> struct ast_format_cap *incoming_call_offer_cap;</span><br><span> struct ast_format_cap *remote;</span><br><span> struct ast_rtp_codecs codecs = AST_RTP_CODECS_NULL_INIT;</span><br><span style="color: hsl(0, 100%, 40%);">- int fmts = 0;</span><br><span> SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));</span><br><span> </span><br><span> </span><br><span>@@ -454,8 +464,7 @@</span><br><span> }</span><br><span> </span><br><span> /* Get the peer's capabilities*/</span><br><span style="color: hsl(0, 100%, 40%);">- get_codecs(session, stream, &codecs, session_media);</span><br><span style="color: hsl(0, 100%, 40%);">- ast_rtp_codecs_payload_formats(&codecs, remote, &fmts);</span><br><span style="color: hsl(120, 100%, 40%);">+ get_codecs(session, stream, &codecs, session_media, remote);</span><br><span> </span><br><span> incoming_call_offer_cap = ast_sip_session_create_joint_call_cap(</span><br><span> session, session_media->type, remote);</span><br><span>@@ -493,7 +502,6 @@</span><br><span> RAII_VAR(struct ast_format_cap *, joint, NULL, ao2_cleanup);</span><br><span> enum ast_media_type media_type = session_media->type;</span><br><span> struct ast_rtp_codecs codecs = AST_RTP_CODECS_NULL_INIT;</span><br><span style="color: hsl(0, 100%, 40%);">- int fmts = 0;</span><br><span> int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&</span><br><span> ast_format_cap_count(session->direct_media_cap);</span><br><span> int dsp_features = 0;</span><br><span>@@ -516,8 +524,7 @@</span><br><span> }</span><br><span> </span><br><span> /* get the capabilities on the peer */</span><br><span style="color: hsl(0, 100%, 40%);">- get_codecs(session, stream, &codecs, session_media);</span><br><span style="color: hsl(0, 100%, 40%);">- ast_rtp_codecs_payload_formats(&codecs, peer, &fmts);</span><br><span style="color: hsl(120, 100%, 40%);">+ get_codecs(session, stream, &codecs, session_media, peer);</span><br><span> </span><br><span> /* get the joint capabilities between peer and endpoint */</span><br><span> ast_format_cap_get_compatible(caps, peer, joint);</span><br><span></span><br></pre><div style="white-space:pre-wrap"></div><p>To view, visit <a href="https://gerrit.asterisk.org/c/asterisk/+/17730">change 17730</a>. 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<div style="display:none"> Gerrit-Project: asterisk </div>
<div style="display:none"> Gerrit-Branch: 19 </div>
<div style="display:none"> Gerrit-Change-Id: I3aabcfed3f379c36654f59c1872c313d0cb57e25 </div>
<div style="display:none"> Gerrit-Change-Number: 17730 </div>
<div style="display:none"> Gerrit-PatchSet: 3 </div>
<div style="display:none"> Gerrit-Owner: Florentin Mayer <f.mayer@commend.com> </div>
<div style="display:none"> Gerrit-Reviewer: Friendly Automation </div>
<div style="display:none"> Gerrit-Reviewer: George Joseph <gjoseph@digium.com> </div>
<div style="display:none"> Gerrit-Reviewer: Joshua Colp <jcolp@sangoma.com> </div>
<div style="display:none"> Gerrit-Reviewer: Kevin Harwell <kharwell@digium.com> </div>
<div style="display:none"> Gerrit-MessageType: merged </div>