res_pjsip_sdp_rtp: Only do hold/unhold on default audio stream.

When examining a stream to determine hold/unhold information we
only care about the default audio stream. Other streams aren't
used for hold/unhold.

ASTERISK-28784

Change-Id: I7a1f10f07822c4aee1f98a38b9628849b578afe4

res/res_pjsip_sdp_rtp.c

@@ -1488,6 +1488,7 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session,
 	/* If ICE support is enabled find all the needed attributes */
 	check_ice_support(session, session_media, stream);
 
+	if (ast_sip_session_is_pending_stream_default(session, asterisk_stream) && media_type == AST_MEDIA_TYPE_AUDIO) {
 		/* Check if incomming SDP is changing the remotely held state */
 		if (ast_sockaddr_isnull(addrs) ||
 			ast_sockaddr_is_any(addrs) ||
@@ -1501,6 +1502,7 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session,
 			session_media->remotely_held = 0;
 			session_media->remotely_held_changed = 1;
 		}
+	}
 
 	if (apply_cap_to_bundled(session_media, session_media_transport, asterisk_stream,
 			set_incoming_call_offer_cap(session, session_media, stream))) {