[Asterisk-code-review] res pjsip sdp rtp.c: Don't alter global addr variable. (asterisk[master])

Anonymous Coward asteriskteam at digium.com
Tue Apr 4 17:53:16 CDT 2017


Anonymous Coward #1000019 has submitted this change and it was merged. ( https://gerrit.asterisk.org/5401 )

Change subject: res_pjsip_sdp_rtp.c: Don't alter global addr variable.
......................................................................


res_pjsip_sdp_rtp.c: Don't alter global addr variable.

* create_rtp(): Fix unexpected alteration of global address_rtp if a
transport is bound to an address.

* create_rtp(): Fix use of uninitialized memory if the endpoint RTP media
address is invalid or the transport has an invalid address.

ASTERISK-26851

Change-Id: Icde42e65164a88913cb5c2601b285eebcff397b7
---
M res/res_pjsip_sdp_rtp.c
1 file changed, 18 insertions(+), 5 deletions(-)

Approvals:
  Mark Michelson: Looks good to me, approved
  George Joseph: Looks good to me, but someone else must approve
  Anonymous Coward #1000019: Verified
  Joshua Colp: Looks good to me, but someone else must approve



diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 21de440..701edc3 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -198,8 +198,16 @@
 	struct ast_sockaddr *media_address =  &address_rtp;
 
 	if (session->endpoint->media.bind_rtp_to_media_address && !ast_strlen_zero(session->endpoint->media.address)) {
-		ast_sockaddr_parse(&temp_media_address, session->endpoint->media.address, 0);
-		media_address = &temp_media_address;
+		if (ast_sockaddr_parse(&temp_media_address, session->endpoint->media.address, 0)) {
+			ast_debug(1, "Endpoint %s: Binding RTP media to %s\n",
+				ast_sorcery_object_get_id(session->endpoint),
+				session->endpoint->media.address);
+			media_address = &temp_media_address;
+		} else {
+			ast_debug(1, "Endpoint %s: RTP media address invalid: %s\n",
+				ast_sorcery_object_get_id(session->endpoint),
+				session->endpoint->media.address);
+		}
 	} else {
 		struct ast_sip_transport *transport =
 			ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport",
@@ -209,9 +217,14 @@
 			char hoststr[PJ_INET6_ADDRSTRLEN];
 
 			pj_sockaddr_print(&transport->state->host, hoststr, sizeof(hoststr), 0);
-			ast_debug(1, "Transport: %s bound to host: %s, using this for media.\n",
-					  session->endpoint->transport, hoststr);
-			ast_sockaddr_parse(media_address, hoststr, 0);
+			if (ast_sockaddr_parse(&temp_media_address, hoststr, 0)) {
+				ast_debug(1, "Transport %s bound to %s: Using it for RTP media.\n",
+					session->endpoint->transport, hoststr);
+				media_address = &temp_media_address;
+			} else {
+				ast_debug(1, "Transport %s bound to %s: Invalid for RTP media.\n",
+					session->endpoint->transport, hoststr);
+			}
 		}
 		ao2_cleanup(transport);
 	}

-- 
To view, visit https://gerrit.asterisk.org/5401
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-MessageType: merged
Gerrit-Change-Id: Icde42e65164a88913cb5c2601b285eebcff397b7
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>



More information about the asterisk-code-review mailing list