kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-10-13 00:04:53 +00:00
Code Issues Packages Projects Releases Wiki Activity

Merge in current pimp_my_sip work, including:

Browse Source 
1. Security events
2. Websocket support
3. Diversion header + redirecting support
4. An anonymous endpoint identifier
5. Inbound extension state subscription support
6. PIDF notify generation
7. One touch recording support (special thanks Sean Bright!)
8. Blind and attended transfer support
9. Automatic inbound registration expiration
10. SRTP support
11. Media offer control dialplan function
12. Connected line support
13. SendText() support
14. Qualify support
15. Inband DTMF detection
16. Call and pickup groups
17. Messaging support

Thanks everyone!

Side note: I'm reminded of the song "How Far We've Come" by Matchbox Twenty.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2013-06-22 14:03:22 +00:00
parent ea03516cb5
commit 77002bc377
42 changed files with 6437 additions and 660 deletions
Download Patch File Download Diff File Expand all files Collapse all files

136
channels/chan_sip.c
View File

@@ -285,8 +285,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "sip/include/config_parser.h"
#include "sip/include/reqresp_parser.h"
#include "sip/include/sip_utils.h"
#include "sip/include/srtp.h"
#include "sip/include/sdp_crypto.h"
#include "asterisk/sdp_srtp.h"
#include "asterisk/ccss.h"
#include "asterisk/xml.h"
#include "sip/include/dialog.h"
@@ -1490,8 +1489,7 @@ static int handle_response_register(struct sip_pvt *p, int resp, const char *res
static void handle_response(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
/*------ SRTP Support -------- */
static int setup_srtp(struct sip_srtp **srtp);
static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct sip_srtp **srtp, const char *a);
static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct ast_sdp_srtp **srtp, const char *a);
/*------ T38 Support --------- */
static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
@@ -5918,7 +5916,7 @@ static void copy_socket_data(struct sip_socket *to_sock, const struct sip_socket
}
/*! \brief Initialize DTLS-SRTP support on an RTP instance */
static int dialog_initialize_dtls_srtp(const struct sip_pvt *dialog, struct ast_rtp_instance *rtp, struct sip_srtp **srtp)
static int dialog_initialize_dtls_srtp(const struct sip_pvt *dialog, struct ast_rtp_instance *rtp, struct ast_sdp_srtp **srtp)
{
struct ast_rtp_engine_dtls *dtls;
@@ -5943,7 +5941,7 @@ static int dialog_initialize_dtls_srtp(const struct sip_pvt *dialog, struct ast_
return -1;
}
if (!(*srtp = sip_srtp_alloc())) {
if (!(*srtp = ast_sdp_srtp_alloc())) {
ast_log(LOG_ERROR, "Failed to create required SRTP structure on RTP instance '%p'\n",
rtp);
return -1;
@@ -6418,17 +6416,17 @@ static int sip_call(struct ast_channel *ast, const char *dest, int timeout)
ast_clear_flag(&p->flags[0], SIP_REINVITE);
}
if (p->rtp && !p->srtp && setup_srtp(&p->srtp) < 0) {
if (p->rtp && !p->srtp && !(p->srtp = ast_sdp_srtp_alloc())) {
ast_log(LOG_WARNING, "SRTP audio setup failed\n");
return -1;
}
if (p->vrtp && !p->vsrtp && setup_srtp(&p->vsrtp) < 0) {
if (p->vrtp && !p->vsrtp && !(p->vsrtp = ast_sdp_srtp_alloc())) {
ast_log(LOG_WARNING, "SRTP video setup failed\n");
return -1;
}
if (p->trtp && !p->tsrtp && setup_srtp(&p->tsrtp) < 0) {
if (p->trtp && !p->tsrtp && !(p->tsrtp = ast_sdp_srtp_alloc())) {
ast_log(LOG_WARNING, "SRTP text setup failed\n");
return -1;
}
@@ -6690,17 +6688,17 @@ void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist)
destroy_msg_headers(p);
if (p->srtp) {
sip_srtp_destroy(p->srtp);
ast_sdp_srtp_destroy(p->srtp);
p->srtp = NULL;
}
if (p->vsrtp) {
sip_srtp_destroy(p->vsrtp);
ast_sdp_srtp_destroy(p->vsrtp);
p->vsrtp = NULL;
}
if (p->tsrtp) {
sip_srtp_destroy(p->tsrtp);
ast_sdp_srtp_destroy(p->tsrtp);
p->tsrtp = NULL;
}
@@ -10154,7 +10152,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
secure_audio = 1;
if (p->srtp) {
ast_set_flag(p->srtp, SRTP_CRYPTO_OFFER_OK);
ast_set_flag(p->srtp, AST_SRTP_CRYPTO_OFFER_OK);
}
} else if (!strcmp(protocol, "RTP/SAVP") || !strcmp(protocol, "RTP/SAVPF")) {
secure_audio = 1;
@@ -10235,8 +10233,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
} else if (!strcmp(protocol, "UDP/TLS/RTP/SAVP") || !strcmp(protocol, "UDP/TLS/RTP/SAVPF")) {
secure_video = 1;
if (p->vsrtp || (p->vsrtp = sip_srtp_alloc())) {
ast_set_flag(p->vsrtp, SRTP_CRYPTO_OFFER_OK);
if (p->vsrtp || (p->vsrtp = ast_sdp_srtp_alloc())) {
ast_set_flag(p->vsrtp, AST_SRTP_CRYPTO_OFFER_OK);
}
} else if (!strcmp(protocol, "RTP/SAVP") || !strcmp(protocol, "RTP/SAVPF")) {
secure_video = 1;
@@ -10516,7 +10514,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
goto process_sdp_cleanup;
}
if (secure_audio && !(p->srtp && (ast_test_flag(p->srtp, SRTP_CRYPTO_OFFER_OK)))) {
if (secure_audio && !(p->srtp && (ast_test_flag(p->srtp, AST_SRTP_CRYPTO_OFFER_OK)))) {
ast_log(LOG_WARNING, "Can't provide secure audio requested in SDP offer\n");
res = -1;
goto process_sdp_cleanup;
@@ -10528,7 +10526,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
goto process_sdp_cleanup;
}
if (secure_video && !(p->vsrtp && (ast_test_flag(p->vsrtp, SRTP_CRYPTO_OFFER_OK)))) {
if (secure_video && !(p->vsrtp && (ast_test_flag(p->vsrtp, AST_SRTP_CRYPTO_OFFER_OK)))) {
ast_log(LOG_WARNING, "Can't provide secure video requested in SDP offer\n");
res = -1;
goto process_sdp_cleanup;
@@ -12993,52 +12991,20 @@ static void get_our_media_address(struct sip_pvt *p, int needvideo, int needtext
}
}
static void get_crypto_attrib(struct sip_pvt *p, struct sip_srtp *srtp, const char **a_crypto)
static char *crypto_get_attrib(struct ast_sdp_srtp *srtp, int dtls_enabled, int default_taglen_32)
{
int taglen = 80;
char *a_crypto;
char *orig_crypto;
/* Set encryption properties */
if (srtp) {
if (!srtp->crypto) {
srtp->crypto = sdp_crypto_setup();
}
if (p->dtls_cfg.enabled) {
/* If DTLS-SRTP is enabled the key details will be pulled from TLS */
return;
}
/* set the key length based on INVITE or settings */
if (ast_test_flag(srtp, SRTP_CRYPTO_TAG_80)) {
taglen = 80;
} else if (ast_test_flag(&p->flags[2], SIP_PAGE3_SRTP_TAG_32) ||
ast_test_flag(srtp, SRTP_CRYPTO_TAG_32)) {
taglen = 32;
}
if (srtp->crypto && (sdp_crypto_offer(srtp->crypto, taglen) >= 0)) {
*a_crypto = sdp_crypto_attrib(srtp->crypto);
}
if (!*a_crypto) {
ast_log(LOG_WARNING, "No SRTP key management enabled\n");
}
if (!srtp) {
return NULL;
}
}
static char *get_sdp_rtp_profile(const struct sip_pvt *p, unsigned int secure, struct ast_rtp_instance *instance)
{
struct ast_rtp_engine_dtls *dtls;
if ((dtls = ast_rtp_instance_get_dtls(instance)) && dtls->active(instance)) {
return ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF) ? "UDP/TLS/RTP/SAVPF" : "UDP/TLS/RTP/SAVP";
} else {
if (ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
return secure ? "RTP/SAVPF" : "RTP/AVPF";
} else {
return secure ? "RTP/SAVP" : "RTP/AVP";
}
orig_crypto = ast_strdupa(ast_sdp_srtp_get_attrib(srtp, dtls_enabled, default_taglen_32));
if (ast_asprintf(&a_crypto, "a=crypto:%s\r\n", orig_crypto) == -1) {
return NULL;
}
return a_crypto;
}
/*! \brief Add Session Description Protocol message
@@ -13079,9 +13045,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
struct ast_str *a_video = ast_str_create(256); /* Attributes for video */
struct ast_str *a_text = ast_str_create(256); /* Attributes for text */
struct ast_str *a_modem = ast_str_alloca(1024); /* Attributes for modem */
const char *a_crypto = NULL;
const char *v_a_crypto = NULL;
const char *t_a_crypto = NULL;
RAII_VAR(char *, a_crypto, NULL, ast_free);
RAII_VAR(char *, v_a_crypto, NULL, ast_free);
RAII_VAR(char *, t_a_crypto, NULL, ast_free);
int x;
struct ast_format tmp_fmt;
@@ -13199,9 +13165,11 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
/* Ok, we need video. Let's add what we need for video and set codecs.
Video is handled differently than audio since we can not transcode. */
if (needvideo) {
get_crypto_attrib(p, p->vsrtp, &v_a_crypto);
v_a_crypto = crypto_get_attrib(p->vsrtp, p->dtls_cfg.enabled,
ast_test_flag(&p->flags[2], SIP_PAGE3_SRTP_TAG_32));
ast_str_append(&m_video, 0, "m=video %d %s", ast_sockaddr_port(&vdest),
get_sdp_rtp_profile(p, v_a_crypto ? 1 : 0, p->vrtp));
ast_sdp_get_rtp_profile(v_a_crypto ? 1 : 0, p->vrtp,
ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)));
/* Build max bitrate string */
if (p->maxcallbitrate)
@@ -13224,9 +13192,11 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
if (needtext) {
if (sipdebug_text)
ast_verbose("Lets set up the text sdp\n");
get_crypto_attrib(p, p->tsrtp, &t_a_crypto);
t_a_crypto = crypto_get_attrib(p->tsrtp, p->dtls_cfg.enabled,
ast_test_flag(&p->flags[2], SIP_PAGE3_SRTP_TAG_32));
ast_str_append(&m_text, 0, "m=text %d %s", ast_sockaddr_port(&tdest),
get_sdp_rtp_profile(p, t_a_crypto ? 1 : 0, p->trtp));
ast_sdp_get_rtp_profile(t_a_crypto ? 1 : 0, p->trtp,
ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)));
if (debug) { /* XXX should I use tdest below ? */
ast_verbose("Text is at %s\n", ast_sockaddr_stringify(&taddr));
}
@@ -13245,9 +13215,11 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
/* We break with the "recommendation" and send our IP, in order that our
peer doesn't have to ast_gethostbyname() us */
get_crypto_attrib(p, p->srtp, &a_crypto);
a_crypto = crypto_get_attrib(p->srtp, p->dtls_cfg.enabled,
ast_test_flag(&p->flags[2], SIP_PAGE3_SRTP_TAG_32));
ast_str_append(&m_audio, 0, "m=audio %d %s", ast_sockaddr_port(&dest),
get_sdp_rtp_profile(p, a_crypto ? 1 : 0, p->rtp));
ast_sdp_get_rtp_profile(a_crypto ? 1 : 0, p->rtp,
ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)));
/* Now, start adding audio codecs. These are added in this order:
- First what was requested by the calling channel
@@ -25767,7 +25739,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str
transmit_response_with_t38_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL)));
} else if ((p->t38.state == T38_DISABLED) || (p->t38.state == T38_REJECTED)) {
/* If this is not a re-invite or something to ignore - it's critical */
if (p->srtp && !ast_test_flag(p->srtp, SRTP_CRYPTO_OFFER_OK)) {
if (p->srtp && !ast_test_flag(p->srtp, AST_SRTP_CRYPTO_OFFER_OK)) {
ast_log(LOG_WARNING, "Target does not support required crypto\n");
transmit_response_reliable(p, "488 Not Acceptable Here (crypto)", req);
} else {
@@ -32791,22 +32763,7 @@ static void sip_send_all_mwi_subscriptions(void)
} while (0));
}
/* SRTP */
static int setup_srtp(struct sip_srtp **srtp)
{
if (!ast_rtp_engine_srtp_is_registered()) {
ast_debug(1, "No SRTP module loaded, can't setup SRTP session.\n");
return -1;
}
if (!(*srtp = sip_srtp_alloc())) { /* Allocate SRTP data structure */
return -1;
}
return 0;
}
static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct sip_srtp **srtp, const char *a)
static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct ast_sdp_srtp **srtp, const char *a)
{
struct ast_rtp_engine_dtls *dtls;
@@ -32819,27 +32776,28 @@ static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struc
if (strncasecmp(a, "crypto:", 7)) {
return FALSE;
}
/* skip "crypto:" */
a += strlen("crypto:");
if (!*srtp) {
if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
ast_log(LOG_WARNING, "Ignoring unexpected crypto attribute in SDP answer\n");
return FALSE;
}
if (setup_srtp(srtp) < 0) {
if (!(*srtp = ast_sdp_srtp_alloc())) {
return FALSE;
}
}
if (!(*srtp)->crypto && !((*srtp)->crypto = sdp_crypto_setup())) {
if (!(*srtp)->crypto && !((*srtp)->crypto = ast_sdp_crypto_alloc())) {
return FALSE;
}
if (sdp_crypto_process((*srtp)->crypto, a, rtp, *srtp) < 0) {
if (ast_sdp_crypto_process(rtp, *srtp, a) < 0) {
return FALSE;
}
ast_set_flag(*srtp, SRTP_CRYPTO_OFFER_OK);
if ((dtls = ast_rtp_instance_get_dtls(rtp))) {
dtls->stop(rtp);
p->dtls_cfg.enabled = 0;