sip-patch-1

sip-patch-1

From 4b0e4fb2f02a29193779c6aa2482329311b813c3 Mon Sep 17 00:00:00 2001
From: Arun Raghavan <arun@asymptotic.io>
Date: Mon, 9 Jan 2023 21:16:48 -0500
Subject: [PATCH 1/3] webrtcbin: Accept SDP without a=mid lines

This should be fine for non-bundle media.
---
 subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c | 4 +---
 subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.c    | 7 +++----
 2 files changed, 4 insertions(+), 7 deletions(-)

diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
index ad36677c85..2551438d8d 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
+++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
@@ -4464,18 +4464,16 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
       if (g_strcmp0 (attr->key, "mid") == 0
           || g_strcmp0 (attr->key, "rtcp-mux") == 0) {
         gst_sdp_media_add_attribute (media, attr->key, attr->value);
         /* FIXME: handle anything we want to keep */
       }
     }
 
     mid = gst_sdp_media_get_attribute_val (media, "mid");
-    /* XXX: not strictly required but a lot of functionality requires a mid */
-    g_assert (mid);
 
     /* set the a=setup: attribute */
     offer_setup = _get_dtls_setup_from_media (offer_media);
     answer_setup = _intersect_dtls_setup (offer_setup);
     if (answer_setup == GST_WEBRTC_DTLS_SETUP_NONE) {
       GST_WARNING_OBJECT (webrtc, "Could not intersect offer setup with "
           "transceiver direction");
       goto rejected;
@@ -4526,17 +4524,17 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
       gint target_pt = -1;
       gint original_target_pt = -1;
       guint target_ssrc = 0;
 
       gst_sdp_media_set_proto (media, "UDP/TLS/RTP/SAVPF");
       offer_caps = _rtp_caps_from_media (offer_media);
 
       if (last_answer && i < gst_sdp_message_medias_len (last_answer)
-          && (rtp_trans = _find_transceiver_for_mid (webrtc, mid))) {
+          && mid && (rtp_trans = _find_transceiver_for_mid (webrtc, mid))) {
         const GstSDPMedia *last_media =
             gst_sdp_message_get_media (last_answer, i);
         const gchar *last_mid =
             gst_sdp_media_get_attribute_val (last_media, "mid");
         GstCaps *current_caps;
 
         /* FIXME: assumes no shenanigans with recycling transceivers */
         g_assert (g_strcmp0 (mid, last_mid) == 0);
diff --git a/subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.c b/subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.c
index 1abd4b1159..82a7c5fc79 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.c
+++ b/subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.c
@@ -293,21 +293,20 @@ validate_sdp (GstWebRTCSignalingState state, SDPSource source,
     return FALSE;*/
   group = gst_sdp_message_get_attribute_val (sdp->sdp, "group");
   is_bundle = group && g_str_has_prefix (group, "BUNDLE");
   if (is_bundle)
     group_members = g_strsplit (&group[6], " ", -1);
 
   for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) {
     const GstSDPMedia *media = gst_sdp_message_get_media (sdp->sdp, i);
-    const gchar *mid;
+    const gchar *mid = NULL;
     gboolean media_in_bundle = FALSE;
-    if (!_media_has_mid (media, i, error))
-      goto fail;
-    mid = gst_sdp_media_get_attribute_val (media, "mid");
+    if (is_bundle && _media_has_mid (media, i, error))
+      mid = gst_sdp_media_get_attribute_val (media, "mid");
     media_in_bundle = is_bundle
         && g_strv_contains ((const gchar **) group_members, mid);
     if (!_media_get_ice_ufrag (sdp->sdp, i)) {
       g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
           "media %u is missing or contains an empty \'ice-ufrag\' attribute",
           i);
       goto fail;
     }
-- 
2.41.0