[Asterisk-code-review] pjsip: Support dual stack automatically. (asterisk[14])

Joshua Colp asteriskteam at digium.com
Mon Sep 26 18:56:38 CDT 2016


Joshua Colp has uploaded a new change for review.

  https://gerrit.asterisk.org/3976

Change subject: pjsip: Support dual stack automatically.
......................................................................

pjsip: Support dual stack automatically.

This change adds support for dual stack automatically. No
configuration is required and the IP address and version
in the SIP messages and SDP will be automatically changed
based on the transport over which the message is being
sent. RTP usage has also been changed to listen on both
IPv4 and IPv6 simultaneously to allow media to flow, and
to allow ICE support on both simultaneously. This also
allows failover between IPv6 and IPv4 to work as expected.

ASTERISK-26309 #close

Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d
---
M CHANGES
M configs/samples/pjsip.conf.sample
M contrib/scripts/sip_to_pjsip/sip_to_pjsip.py
M res/res_pjsip_multihomed.c
M res/res_pjsip_sdp_rtp.c
M res/res_pjsip_t38.c
6 files changed, 143 insertions(+), 63 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/76/3976/1

diff --git a/CHANGES b/CHANGES
index 350c00c..eb149c0 100644
--- a/CHANGES
+++ b/CHANGES
@@ -71,6 +71,11 @@
    Note: The caller-id and redirecting number strings obtained from incoming
    SIP URI user fields are now always truncated at the first semicolon.
 
+ * Automatic dual stack support is now implemented. Depending on DNS resolution
+   and the transport used for sending a message the SIP signaling and SDP will
+   be updated with the correct IP address and protocol version. This means that
+   the rtp_ipv6 and t38_udptl_ipv6 options no longer have any effect.
+
 app_confbridge
 ------------------
   * Some sounds played into the bridge are played asynchronously. This, for
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index 8cd732a..eda8022 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -112,9 +112,6 @@
 ; the prefix "external_" will only apply to communication with addresses
 ; outside the range set with "local_net=".
 ;
-; IPv6: For endpoints using IPv6, remember to set "rtp_ipv6=yes" so that the RTP
-; engine will also be able to bind to an IPv6 address.
-;
 ; You can have more than one of any type of transport, as long as it doesn't
 ; use the same resources (bind address, port, etc) as the others.
 
@@ -294,8 +291,6 @@
 ; If using the TLS enabled transport, you may want the "media_encryption=sdes"
 ; option to additionally enable SRTP, though they are not mutually inclusive.
 ;
-; Use the "rtp_ipv6=yes" option if you want to utilize RTP over an ipv6 transport.
-;
 ; If this endpoint were remote, and it was using a transport configured for NAT
 ; then you likely want to use "direct_media=no" to prevent audio issues.
 
@@ -315,7 +310,6 @@
 ;transport=transport-tls
 ;media_encryption=sdes
 ;transport=transport-udp-ipv6
-;rtp_ipv6=yes
 ;transport=transport-udp-nat
 ;direct_media=no
 ;
@@ -646,7 +640,6 @@
                         ; must be provided (default: "")
 ;rewrite_contact=no     ; Allow Contact header to be rewritten with the source
                         ; IP address port (default: "no")
-;rtp_ipv6=no    ; Allow use of IPv6 for RTP traffic (default: "no")
 ;rtp_symmetric=no       ; Enforce that RTP must be symmetric (default: "no")
 ;send_diversion=yes     ; Send the Diversion header conveying the diversion
                         ; information to the called user agent (default: "yes")
@@ -699,8 +692,6 @@
                         ; (default: "0")
 ;t38_udptl_nat=no       ; Whether NAT support is enabled on UDPTL sessions
                         ; (default: "no")
-;t38_udptl_ipv6=no      ; Whether IPv6 is used for UDPTL Sessions (default:
-                        ; "no")
 ;tone_zone=     ; Set which country s indications to use for channels created
                 ; for this endpoint (default: "")
 ;language=      ; Set the default language to use for channels created for this
diff --git a/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py b/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py
index 40e9354..98a5e95 100755
--- a/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py
+++ b/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py
@@ -374,7 +374,7 @@
 ###############################################################################
 
 # options in pjsip.conf on an endpoint that have no sip.conf equivalent:
-# type, rtp_ipv6, 100rel, trust_id_outbound, aggregate_mwi,
+# type, 100rel, trust_id_outbound, aggregate_mwi,
 # connected_line_method
 
 # known sip.conf peer keys that can be mapped to a pjsip.conf section/key
diff --git a/res/res_pjsip_multihomed.c b/res/res_pjsip_multihomed.c
index 5deeb92..2708361 100644
--- a/res/res_pjsip_multihomed.c
+++ b/res/res_pjsip_multihomed.c
@@ -1,7 +1,7 @@
 /*
  * Asterisk -- An open source telephony toolkit.
  *
- * Copyright (C) 2014, Digium, Inc.
+ * Copyright (C) 2014-2016, Digium, Inc.
  *
  * Joshua Colp <jcolp at digium.com>
  *
@@ -19,6 +19,7 @@
 /*** MODULEINFO
 	<depend>pjproject</depend>
 	<depend>res_pjsip</depend>
+	<depend>res_pjsip_session</depend>
 	<support_level>core</support_level>
  ***/
 
@@ -28,7 +29,72 @@
 #include <pjsip_ua.h>
 
 #include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
 #include "asterisk/module.h"
+
+#define MOD_DATA_RESTRICTIONS "restrictions"
+
+static pj_status_t multihomed_on_tx_message(pjsip_tx_data *tdata);
+
+/*! \brief Outgoing message modification restrictions */
+struct multihomed_message_restrictions {
+	/*! \brief Disallow modification of the From domain */
+	unsigned int disallow_from_domain_modification;
+};
+
+static pjsip_module multihomed_module = {
+	.name = { "Multihomed Routing", 18 },
+	.id = -1,
+	.priority = PJSIP_MOD_PRIORITY_TSX_LAYER - 1,
+	.on_tx_request = multihomed_on_tx_message,
+	.on_tx_response = multihomed_on_tx_message,
+};
+
+/*! \brief Helper function to get (or allocate if not already present) restrictions on a message */
+static struct multihomed_message_restrictions *multihomed_get_restrictions(pjsip_tx_data *tdata)
+{
+	struct multihomed_message_restrictions *restrictions;
+
+	restrictions = ast_sip_mod_data_get(tdata->mod_data, multihomed_module.id, MOD_DATA_RESTRICTIONS);
+	if (restrictions) {
+		return restrictions;
+	}
+
+	restrictions = PJ_POOL_ALLOC_T(tdata->pool, struct multihomed_message_restrictions);
+	ast_sip_mod_data_set(tdata->pool, tdata->mod_data, multihomed_module.id, MOD_DATA_RESTRICTIONS, restrictions);
+
+	return restrictions;
+}
+
+/*! \brief Callback invoked on non-session outgoing messages */
+static void multihomed_outgoing_message(struct ast_sip_endpoint *endpoint, struct ast_sip_contact *contact, struct pjsip_tx_data *tdata)
+{
+	struct multihomed_message_restrictions *restrictions = multihomed_get_restrictions(tdata);
+
+	restrictions->disallow_from_domain_modification = !ast_strlen_zero(endpoint->fromdomain);
+}
+
+/*! \brief PJSIP Supplement for tagging messages with restrictions */
+static struct ast_sip_supplement multihomed_supplement = {
+	.priority = AST_SIP_SUPPLEMENT_PRIORITY_FIRST,
+	.outgoing_request = multihomed_outgoing_message,
+	.outgoing_response = multihomed_outgoing_message,
+};
+
+/*! \brief Callback invoked on session outgoing messages */
+static void multihomed_session_outgoing_message(struct ast_sip_session *session, struct pjsip_tx_data *tdata)
+{
+	struct multihomed_message_restrictions *restrictions = multihomed_get_restrictions(tdata);
+
+	restrictions->disallow_from_domain_modification = !ast_strlen_zero(session->endpoint->fromdomain);
+}
+
+/*! \brief PJSIP Session Supplement for tagging messages with restrictions */
+static struct ast_sip_session_supplement multihomed_session_supplement = {
+	.priority = 1,
+	.outgoing_request = multihomed_session_outgoing_message,
+	.outgoing_response = multihomed_session_outgoing_message,
+};
 
 /*! \brief Helper function which returns a UDP transport bound to the given address and port */
 static pjsip_transport *multihomed_get_udp_transport(pj_str_t *address, int port)
@@ -59,6 +125,21 @@
 	return sip_transport;
 }
 
+/*! \brief Helper function which determines if a transport is bound to any */
+static int multihomed_bound_any(pjsip_transport *transport)
+{
+	pj_uint32_t loop6[4] = {0, 0, 0, 0};
+
+	if ((transport->local_addr.addr.sa_family == pj_AF_INET() &&
+		transport->local_addr.ipv4.sin_addr.s_addr == PJ_INADDR_ANY) ||
+		(transport->local_addr.addr.sa_family == pj_AF_INET6() &&
+		!pj_memcmp(&transport->local_addr.ipv6.sin6_addr, loop6, sizeof(loop6)))) {
+		return 1;
+	}
+
+	return 0;
+}
+
 /*! \brief Helper function which determines if the address within SDP should be rewritten */
 static int multihomed_rewrite_sdp(struct pjmedia_sdp_session *sdp)
 {
@@ -77,26 +158,13 @@
 	return 0;
 }
 
-/*! \brief Helper function which determines if a transport is bound to any */
-static int multihomed_bound_any(pjsip_transport *transport)
-{
-	pj_uint32_t loop6[4] = {0, 0, 0, 0};
-
-	if ((transport->local_addr.addr.sa_family == pj_AF_INET() &&
-		transport->local_addr.ipv4.sin_addr.s_addr == PJ_INADDR_ANY) ||
-		(transport->local_addr.addr.sa_family == pj_AF_INET6() &&
-		!pj_memcmp(&transport->local_addr.ipv6.sin6_addr, loop6, sizeof(loop6)))) {
-		return 1;
-	}
-
-	return 0;
-}
-
 static pj_status_t multihomed_on_tx_message(pjsip_tx_data *tdata)
 {
+	struct multihomed_message_restrictions *restrictions = ast_sip_mod_data_get(tdata->mod_data, multihomed_module.id, MOD_DATA_RESTRICTIONS);
 	pjsip_tpmgr_fla2_param prm;
 	pjsip_cseq_hdr *cseq;
 	pjsip_via_hdr *via;
+	pjsip_fromto_hdr *from;
 
 	/* Use the destination information to determine what local interface this message will go out on */
 	pjsip_tpmgr_fla2_param_default(&prm);
@@ -153,6 +221,13 @@
 			ast_debug(4, "Re-wrote Contact URI host/port to %.*s:%d\n",
 				(int)pj_strlen(&uri->host), pj_strbuf(&uri->host), uri->port);
 
+			if (tdata->tp_info.transport->key.type == PJSIP_TRANSPORT_UDP ||
+				tdata->tp_info.transport->key.type == PJSIP_TRANSPORT_UDP6) {
+				uri->transport_param.slen = 0;
+			} else {
+				pj_strdup2(tdata->pool, &uri->transport_param, pjsip_transport_get_type_name(tdata->tp_info.transport->key.type));
+			}
+
 			pjsip_tx_data_invalidate_msg(tdata);
 		}
 	}
@@ -164,17 +239,36 @@
 		pjsip_tx_data_invalidate_msg(tdata);
 	}
 
+	if (tdata->msg->type == PJSIP_REQUEST_MSG && (from = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_FROM, NULL)) &&
+		(restrictions && !restrictions->disallow_from_domain_modification)) {
+		pjsip_name_addr *id_name_addr = (pjsip_name_addr *)from->uri;
+		pjsip_sip_uri *uri = pjsip_uri_get_uri(id_name_addr);
+
+		pj_strassign(&uri->host, &prm.ret_addr);
+
+		pjsip_tx_data_invalidate_msg(tdata);
+	}
+
 	/* Update the SDP if it is present */
 	if (tdata->msg->body && ast_sip_is_content_type(&tdata->msg->body->content_type, "application", "sdp") &&
 		multihomed_rewrite_sdp(tdata->msg->body->data)) {
 		struct pjmedia_sdp_session *sdp = tdata->msg->body->data;
+		static const pj_str_t STR_IP4 = { "IP4", 3 };
+		static const pj_str_t STR_IP6 = { "IP6", 3 };
+		pj_str_t STR_IP;
 		int stream;
 
+		STR_IP = tdata->tp_info.transport->key.type & PJSIP_TRANSPORT_IPV6 ? STR_IP6 : STR_IP4;
+
+		pj_strassign(&sdp->origin.addr, &prm.ret_addr);
+		sdp->origin.addr_type = STR_IP;
 		pj_strassign(&sdp->conn->addr, &prm.ret_addr);
+		sdp->conn->addr_type = STR_IP;
 
 		for (stream = 0; stream < sdp->media_count; ++stream) {
 			if (sdp->media[stream]->conn) {
 				pj_strassign(&sdp->media[stream]->conn->addr, &prm.ret_addr);
+				sdp->media[stream]->conn->addr_type = STR_IP;
 			}
 		}
 
@@ -184,40 +278,39 @@
 	return PJ_SUCCESS;
 }
 
-static pjsip_module multihomed_module = {
-	.name = { "Multihomed Routing", 18 },
-	.id = -1,
-	.priority = PJSIP_MOD_PRIORITY_TSX_LAYER - 1,
-	.on_tx_request = multihomed_on_tx_message,
-	.on_tx_response = multihomed_on_tx_message,
-};
-
 static int unload_module(void)
 {
 	ast_sip_unregister_service(&multihomed_module);
+	ast_sip_unregister_supplement(&multihomed_supplement);
+	ast_sip_session_unregister_supplement(&multihomed_session_supplement);
 	return 0;
 }
 
 static int load_module(void)
 {
-	char hostname[MAXHOSTNAMELEN] = "";
-
 	CHECK_PJSIP_MODULE_LOADED();
 
-	if (!gethostname(hostname, sizeof(hostname) - 1)) {
-		ast_verb(2, "Performing DNS resolution of local hostname '%s' to get local IPv4 and IPv6 address\n",
-			hostname);
+	if (ast_sip_session_register_supplement(&multihomed_session_supplement)) {
+		ast_log(LOG_ERROR, "Could not register multihomed session supplement for outgoing requests\n");
+		return AST_MODULE_LOAD_FAILURE;
+	}
+
+	if (ast_sip_register_supplement(&multihomed_supplement)) {
+		ast_log(LOG_ERROR, "Could not register multihomed supplement for outgoing requests\n");
+		unload_module();
+		return AST_MODULE_LOAD_FAILURE;
 	}
 
 	if (ast_sip_register_service(&multihomed_module)) {
 		ast_log(LOG_ERROR, "Could not register multihomed module for incoming and outgoing requests\n");
+		unload_module();
 		return AST_MODULE_LOAD_FAILURE;
 	}
 
 	return AST_MODULE_LOAD_SUCCESS;
 }
 
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Multihomed Routing Support",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Multihomed Routing and Dual Stack Support",
 	.support_level = AST_MODULE_SUPPORT_CORE,
 	.load = load_module,
 	.unload = unload_module,
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 6610ef1..7937972 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -59,11 +59,8 @@
 /*! \brief Scheduler for RTCP purposes */
 static struct ast_sched_context *sched;
 
-/*! \brief Address for IPv4 RTP */
-static struct ast_sockaddr address_ipv4;
-
-/*! \brief Address for IPv6 RTP */
-static struct ast_sockaddr address_ipv6;
+/*! \brief Address for RTP */
+static struct ast_sockaddr address_rtp;
 
 static const char STR_AUDIO[] = "audio";
 static const int FD_AUDIO = 0;
@@ -173,11 +170,11 @@
 }
 
 /*! \brief Internal function which creates an RTP instance */
-static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_media *session_media, unsigned int ipv6)
+static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_media *session_media)
 {
 	struct ast_rtp_engine_ice *ice;
 	struct ast_sockaddr temp_media_address;
-	struct ast_sockaddr *media_address =  ipv6 ? &address_ipv6 : &address_ipv4;
+	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);
@@ -898,7 +895,7 @@
 	}
 
 	/* Using the connection information create an appropriate RTP instance */
-	if (!session_media->rtp && create_rtp(session, session_media, ast_sockaddr_is_ipv6(addrs))) {
+	if (!session_media->rtp && create_rtp(session, session_media)) {
 		return -1;
 	}
 
@@ -1050,7 +1047,6 @@
 	pj_pool_t *pool = session->inv_session->pool_prov;
 	static const pj_str_t STR_IN = { "IN", 2 };
 	static const pj_str_t STR_IP4 = { "IP4", 3};
-	static const pj_str_t STR_IP6 = { "IP6", 3};
 	static const pj_str_t STR_SENDRECV = { "sendrecv", 8 };
 	static const pj_str_t STR_SENDONLY = { "sendonly", 8 };
 	pjmedia_sdp_media *media;
@@ -1074,7 +1070,7 @@
 	    (!use_override_prefs && !ast_format_cap_has_type(session->endpoint->media.codecs, media_type))) {
 		/* If no type formats are configured don't add a stream */
 		return 0;
-	} else if (!session_media->rtp && create_rtp(session, session_media, session->endpoint->media.rtp.ipv6)) {
+	} else if (!session_media->rtp && create_rtp(session, session_media)) {
 		return -1;
 	}
 
@@ -1115,7 +1111,8 @@
 	}
 
 	media->conn->net_type = STR_IN;
-	media->conn->addr_type = session->endpoint->media.rtp.ipv6 ? STR_IP6 : STR_IP4;
+	/* Connection information will be updated by the multihomed module */
+	media->conn->addr_type = STR_IP4;
 	pj_strdup2(pool, &media->conn->addr, hostip);
 	ast_rtp_instance_get_local_address(session_media->rtp, &addr);
 	media->desc.port = direct_media_enabled ? ast_sockaddr_port(&session_media->direct_media_addr) : (pj_uint16_t) ast_sockaddr_port(&addr);
@@ -1252,7 +1249,7 @@
 	}
 
 	/* Create an RTP instance if need be */
-	if (!session_media->rtp && create_rtp(session, session_media, session->endpoint->media.rtp.ipv6)) {
+	if (!session_media->rtp && create_rtp(session, session_media)) {
 		return -1;
 	}
 
@@ -1488,8 +1485,7 @@
 {
 	CHECK_PJSIP_SESSION_MODULE_LOADED();
 
-	ast_sockaddr_parse(&address_ipv4, "0.0.0.0", 0);
-	ast_sockaddr_parse(&address_ipv6, "::", 0);
+	ast_sockaddr_parse(&address_rtp, "::", 0);
 
 	if (!(sched = ast_sched_context_create())) {
 		ast_log(LOG_ERROR, "Unable to create scheduler context.\n");
diff --git a/res/res_pjsip_t38.c b/res/res_pjsip_t38.c
index 150336a..b052cc4 100644
--- a/res/res_pjsip_t38.c
+++ b/res/res_pjsip_t38.c
@@ -51,11 +51,8 @@
 /*! \brief The number of seconds after receiving a T.38 re-invite before automatically rejecting it */
 #define T38_AUTOMATIC_REJECTION_SECONDS 5
 
-/*! \brief Address for IPv4 UDPTL */
-static struct ast_sockaddr address_ipv4;
-
-/*! \brief Address for IPv6 UDPTL */
-static struct ast_sockaddr address_ipv6;
+/*! \brief Address for UDPTL */
+static struct ast_sockaddr address;
 
 /*! \brief T.38 state information */
 struct t38_state {
@@ -259,8 +256,7 @@
 		return 0;
 	}
 
-	if (!(session_media->udptl = ast_udptl_new_with_bindaddr(NULL, NULL, 0,
-		session->endpoint->media.t38.ipv6 ? &address_ipv6 : &address_ipv4))) {
+	if (!(session_media->udptl = ast_udptl_new_with_bindaddr(NULL, NULL, 0, &address))) {
 		return -1;
 	}
 
@@ -922,8 +918,7 @@
 {
 	CHECK_PJSIP_SESSION_MODULE_LOADED();
 
-	ast_sockaddr_parse(&address_ipv4, "0.0.0.0", 0);
-	ast_sockaddr_parse(&address_ipv6, "::", 0);
+	ast_sockaddr_parse(&address, "::", 0);
 
 	if (ast_sip_session_register_supplement(&t38_supplement)) {
 		ast_log(LOG_ERROR, "Unable to register T.38 session supplement\n");

-- 
To view, visit https://gerrit.asterisk.org/3976
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-MessageType: newchange
Gerrit-Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 14
Gerrit-Owner: Joshua Colp <jcolp at digium.com>



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