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res_rtp_asterisk: enable rtcp & QOS stats on native bridge

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Asterisk wasn't generating or forwarding RTCP packets when native
bridge was activated.  Also the stats weren't available via
CHANNEL(qos). Now the RTCP stats are always calculated.

ASTERISK-27158 #close

Change-Id: I46fb8f61c95e836b9d2dda6054b0cf205c16037b
This commit is contained in:
Torrey Searle
2017-07-26 16:17:02 +02:00
committed by Joshua Colp
parent 6dd12aaacd
commit 3608f96ea3
1 changed files with 22 additions and 10 deletions
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32
res/res_rtp_asterisk.c
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@@ -4854,9 +4854,6 @@ static int bridge_p2p_rtp_write(struct ast_rtp_instance *instance,
return -1;
}
rtp->rxcount++;
rtp->rxoctetcount += (len - hdrlen);
/* 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 */
if (ast_rtp_codecs_find_payload_code(ast_rtp_instance_get_codecs(instance1), bridged_payload) == -1) {
ast_debug(1, "Unsupported payload type received \n");
@@ -5092,13 +5089,6 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
}
}
/* If we are directly bridged to another instance send the audio directly out */
instance1 = ast_rtp_instance_get_bridged(instance);
if (instance1
&& !bridge_p2p_rtp_write(instance, instance1, rtpheader, res, hdrlen)) {
return &ast_null_frame;
}
/* If the version is not what we expected by this point then just drop the packet */
if (version != 2) {
return &ast_null_frame;
@@ -5203,6 +5193,28 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
}
/* If we are directly bridged to another instance send the audio directly out,
* but only after updating core information about the received traffic so that
* outgoing RTCP reflects it.
*/
instance1 = ast_rtp_instance_get_bridged(instance);
if (instance1
&& !bridge_p2p_rtp_write(instance, instance1, rtpheader, res, hdrlen)) {
struct timeval rxtime;
struct ast_frame *f;
/* Update statistics for jitter so they are correct in RTCP */
calc_rxstamp(&rxtime, rtp, timestamp, mark);
/* When doing P2P we don't need to raise any frames about SSRC change to the core */
while ((f = AST_LIST_REMOVE_HEAD(&frames, frame_list)) != NULL) {
ast_frfree(f);
}
return &ast_null_frame;
}
if (rtp_debug_test_addr(&addr)) {
ast_verbose("Got RTP packet from %s (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6d)\n",
ast_sockaddr_stringify(&addr),