[Asterisk-code-review] chan sip: Fix improper RTP framing on outgoing calls (asterisk[master])

[Jenkins2]

Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/8453 )

Change subject: chan_sip: Fix improper RTP framing on outgoing calls
......................................................................

chan_sip: Fix improper RTP framing on outgoing calls

The "ptime" SDP parameter received in a SIP response was not honoured.
Moreover, in the abscence of this "ptime" parameter, locally configured
framing was lost during response processing.

This patch systematically stores the framing information in the
ast_rtp_codecs structure, taking it from the response or from the
configuration as appropriate.

ASTERISK-27674

Change-Id: I828a6a98d27a45a8afd07236a2bd0aa3cbd3fb2c
---
M channels/chan_sip.c
1 file changed, 9 insertions(+), 5 deletions(-)

Approvals:
  Richard Mudgett: Looks good to me, but someone else must approve
  Joshua Colp: Looks good to me, approved
  Jenkins2: Approved for Submit



diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 8570163..f0cc2a6 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10964,22 +10964,25 @@
 	if (portno != -1 || vportno != -1 || tportno != -1) {
 		/* We are now ready to change the sip session and RTP structures with the offered codecs, since
 		   they are acceptable */
+		unsigned int framing;
 		ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN);
 		ast_format_cap_append_from_cap(p->jointcaps, newjointcapability, AST_MEDIA_TYPE_UNKNOWN); /* Our joint codec profile for this call */
 		ast_format_cap_remove_by_type(p->peercaps, AST_MEDIA_TYPE_UNKNOWN);
 		ast_format_cap_append_from_cap(p->peercaps, newpeercapability, AST_MEDIA_TYPE_UNKNOWN); /* The other side's capability in latest offer */
 		p->jointnoncodeccapability = newnoncodeccapability;     /* DTMF capabilities */
 
+		tmp_fmt = ast_format_cap_get_format(p->jointcaps, 0);
+		framing = ast_format_cap_get_format_framing(p->jointcaps, tmp_fmt);
 		/* respond with single most preferred joint codec, limiting the other side's choice */
 		if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) {
-			unsigned int framing;
-
-			tmp_fmt = ast_format_cap_get_format(p->jointcaps, 0);
-			framing = ast_format_cap_get_format_framing(p->jointcaps, tmp_fmt);
 			ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN);
 			ast_format_cap_append(p->jointcaps, tmp_fmt, framing);
-			ao2_ref(tmp_fmt, -1);
 		}
+		if (!ast_rtp_codecs_get_framing(&newaudiortp)) {
+			/* Peer did not force us to use a specific framing, so use our own */
+			ast_rtp_codecs_set_framing(&newaudiortp, framing);
+		}
+		ao2_ref(tmp_fmt, -1);
 	}
 
 	/* Setup audio address and port */
@@ -11488,6 +11491,7 @@
 		if (framing && p->autoframing) {
 			ast_debug(1, "Setting framing to %ld\n", framing);
 			ast_format_cap_set_framing(p->caps, framing);
+			ast_rtp_codecs_set_framing(newaudiortp, framing);
 		}
 		found = TRUE;
 	} else if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {

-- 
To view, visit https://gerrit.asterisk.org/8453
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Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-MessageType: merged
Gerrit-Change-Id: I828a6a98d27a45a8afd07236a2bd0aa3cbd3fb2c
Gerrit-Change-Number: 8453
Gerrit-PatchSet: 1
Gerrit-Owner: Jean Aunis - Prescom <jean.aunis at prescom.fr>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
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