struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED;
- int srccodec;
+ int srccodec, destcodec;
/* Lock channels */
ast_channel_lock(dest);
audio_src_res = srcpr->get_rtp_info(src, &srcp);
video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
+ /* Ensure we have at least one matching codec */
+ if (srcpr->get_codec)
+ srccodec = srcpr->get_codec(src);
+ else
+ srccodec = 0;
+ if (destpr->get_codec)
+ destcodec = destpr->get_codec(dest);
+ else
+ destcodec = 0;
+
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
- if (audio_dest_res != AST_RTP_TRY_NATIVE || audio_src_res != AST_RTP_TRY_NATIVE) {
+ if (audio_dest_res != AST_RTP_TRY_NATIVE || audio_src_res != AST_RTP_TRY_NATIVE || !(srccodec & destcodec)) {
/* Somebody doesn't want to play... */
ast_channel_unlock(dest);
ast_channel_unlock(src);
ast_rtp_pt_copy(destp, srcp);
if (vdestp && vsrcp)
ast_rtp_pt_copy(vdestp, vsrcp);
- if (srcpr->get_codec)
- srccodec = srcpr->get_codec(src);
- else
- srccodec = 0;
if (media) {
/* Bridge early */
if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))