From: Jean Aunis <jean.aunis@prescom.fr>
Date: Wed, 14 Feb 2018 13:33:18 +0000 (+0100)
Subject: chan_sip: Fix improper RTP framing on outgoing calls
X-Git-Tag: 13.21.0-rc1~83^2
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=a35a654a523c3de5e0ae658993ccf0de7e6617fb;p=thirdparty%2Fasterisk.git

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
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 0a5caecb96..38cdd53750 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10912,22 +10912,25 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 	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 */
@@ -11436,6 +11439,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
 		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) {