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Merge "res_rtp_asterisk: enable rtcp & QOS stats on native bridge" into 13
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@@ -4855,9 +4855,6 @@ static int bridge_p2p_rtp_write(struct ast_rtp_instance *instance,
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return -1;
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}
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rtp->rxcount++;
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rtp->rxoctetcount += (len - hdrlen);
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/* If the payload coming in is not one of the negotiated ones then send it to the core, this will cause formats to change and the bridge to break */
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if (ast_rtp_codecs_find_payload_code(ast_rtp_instance_get_codecs(instance1), bridged_payload) == -1) {
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ast_debug(1, "Unsupported payload type received \n");
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@@ -5110,13 +5107,6 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
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}
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}
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/* If we are directly bridged to another instance send the audio directly out */
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instance1 = ast_rtp_instance_get_bridged(instance);
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if (instance1
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&& !bridge_p2p_rtp_write(instance, instance1, rtpheader, res, hdrlen)) {
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return &ast_null_frame;
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}
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/* If the version is not what we expected by this point then just drop the packet */
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if (version != 2) {
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return &ast_null_frame;
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@@ -5221,6 +5211,28 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
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rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
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}
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/* If we are directly bridged to another instance send the audio directly out,
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* but only after updating core information about the received traffic so that
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* outgoing RTCP reflects it.
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*/
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instance1 = ast_rtp_instance_get_bridged(instance);
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if (instance1
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&& !bridge_p2p_rtp_write(instance, instance1, rtpheader, res, hdrlen)) {
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struct timeval rxtime;
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struct ast_frame *f;
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/* Update statistics for jitter so they are correct in RTCP */
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calc_rxstamp(&rxtime, rtp, timestamp, mark);
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/* When doing P2P we don't need to raise any frames about SSRC change to the core */
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while ((f = AST_LIST_REMOVE_HEAD(&frames, frame_list)) != NULL) {
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ast_frfree(f);
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}
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return &ast_null_frame;
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}
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if (rtp_debug_test_addr(&addr)) {
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ast_verbose("Got RTP packet from %s (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6d)\n",
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ast_sockaddr_stringify(&addr),
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