diff options
| -rw-r--r-- | CHANGES | 12 | ||||
| -rw-r--r-- | channels/chan_sip.c | 89 | ||||
| -rw-r--r-- | channels/sip/config_parser.c | 118 | ||||
| -rw-r--r-- | channels/sip/include/config_parser.h | 12 | ||||
| -rw-r--r-- | channels/sip/include/sip.h | 5 | ||||
| -rw-r--r-- | channels/sip/include/sip_utils.h | 6 | ||||
| -rw-r--r-- | channels/sip/utils.c | 45 | ||||
| -rw-r--r-- | configs/sip.conf.sample | 29 |
8 files changed, 286 insertions, 30 deletions
@@ -59,6 +59,12 @@ SIP Changes callbackextension options, incoming requests that are matched by address will be matched to the peer with the matching callbackextension if it is available. + * NAT settings are now a combinable list of options. The equivalent of the + deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before. + * Two new NAT options, auto_force_rport and auto_comedia, have been added + which set the force_rport and comedia options automatically if Asterisk + detects that an incoming SIP request crossed a NAT after being sent by + the remote endpoint. Chan_local changes ------------------ @@ -125,6 +131,12 @@ AMI (Asterisk Manager Interface) changes if the i(variable) option is used. StopMixMonitor will accept MixMonitorID as on option to close specific MixMonitors. + * The SIPshowpeer manager action response field "SIP-Forcerport" has been updated + to include information about peers configured with nat=auto_force_rport by + returning "A" if auto_force_rport is set and nat is detected, and "a" if it is + set and nat is not detected. "Y" and "N" are still returned if auto_force_rport + is not enabled. + FAX changes ----------- * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan diff --git a/channels/chan_sip.c b/channels/chan_sip.c index a96d4eaf0..f138bd474 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -15276,6 +15276,21 @@ static enum check_auth_result register_verify(struct sip_pvt *p, struct ast_sock ast_log(LOG_ERROR, "Peer '%s' is trying to register, but not configured as host=dynamic\n", peer->name); res = AUTH_PEER_NOT_DYNAMIC; } else { + if (ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) { + if (p->natdetected) { + ast_set_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT); + } else { + ast_clear_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT); + } + } + if (ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) { + if (p->natdetected) { + ast_set_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP); + } else { + ast_clear_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP); + } + } + ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_NAT_FORCE_RPORT); if (!(res = check_auth(p, req, peer->name, peer->secret, peer->md5secret, SIP_REGISTER, uri2, XMIT_UNRELIABLE, req->ignore))) { if (sip_cancel_destroy(p)) @@ -16354,6 +16369,30 @@ static void check_via(struct sip_pvt *p, struct sip_request *req) ast_sockaddr_set_port(&p->sa, port != 0 ? port : STANDARD_SIP_PORT); + /* Check and see if the requesting UA is likely to be behind a NAT. If they are, set the + * natdetected flag so that later, peers with nat=auto_* can use the value. Also + * set the flags so that Asterisk responds identically whether or not a peer exists + * so as not to leak peer name information. */ + if (ast_sockaddr_cmp(&tmp, &p->recv)) { + char *tmp_str = ast_strdupa(ast_sockaddr_stringify(&tmp)); + ast_debug(3, "NAT detected for %s / %s\n", tmp_str, ast_sockaddr_stringify(&p->recv)); + p->natdetected = 1; + if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) { + ast_set_flag(&p->flags[0], SIP_NAT_FORCE_RPORT); + } + if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) { + ast_set_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP); + } + } else { + p->natdetected = 0; + if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) { + ast_clear_flag(&p->flags[0], SIP_NAT_FORCE_RPORT); + } + if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) { + ast_clear_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP); + } + } + if (sip_debug_test_pvt(p)) { ast_verbose("Sending to %s (%s)\n", ast_sockaddr_stringify(sip_real_dst(p)), @@ -17437,7 +17476,9 @@ static char *_sip_show_peers(int fd, int *total, struct mansession *s, const str snprintf(srch, sizeof(srch), FORMAT2, name, tmp_host, peer->host_dynamic ? " D " : " ", /* Dynamic or not? */ - ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? " N " : " ", /* NAT=yes? */ + ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT) ? + ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? " A " : " a " : + ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? " N " : " ", /* NAT=yes? */ peer->ha ? " A " : " ", /* permit/deny */ tmp_port, status, peer->description ? peer->description : "", @@ -17447,7 +17488,9 @@ static char *_sip_show_peers(int fd, int *total, struct mansession *s, const str ast_cli(fd, FORMAT2, name, tmp_host, peer->host_dynamic ? " D " : " ", /* Dynamic or not? */ - ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? " N " : " ", /* NAT=yes? */ + ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT) ? + ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? " A " : " a " : + ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? " N " : " ", /* NAT=yes? */ peer->ha ? " A " : " ", /* permit/deny */ tmp_port, status, peer->description ? peer->description : "", @@ -17462,7 +17505,10 @@ static char *_sip_show_peers(int fd, int *total, struct mansession *s, const str "IPaddress: %s\r\n" "IPport: %s\r\n" "Dynamic: %s\r\n" + "AutoForcerport: %s\r\n" "Forcerport: %s\r\n" + "AutoComedia: %s\r\n" + "Comedia: %s\r\n" "VideoSupport: %s\r\n" "TextSupport: %s\r\n" "ACL: %s\r\n" @@ -17474,7 +17520,10 @@ static char *_sip_show_peers(int fd, int *total, struct mansession *s, const str ast_sockaddr_isnull(&peer->addr) ? "-none-" : tmp_host, ast_sockaddr_isnull(&peer->addr) ? "0" : tmp_port, peer->host_dynamic ? "yes" : "no", /* Dynamic or not? */ + ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT) ? "yes" : "no", ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? "yes" : "no", /* NAT=yes? */ + ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA) ? "yes" : "no", + ast_test_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP) ? "yes" : "no", ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "yes" : "no", /* VIDEOSUPPORT=yes? */ ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "yes" : "no", /* TEXTSUPPORT=yes? */ peer->ha ? "yes" : "no", /* permit/deny */ @@ -18142,7 +18191,8 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct ast_cli(fd, " MaxCallBR : %d kbps\n", peer->maxcallbitrate); ast_cli(fd, " Expire : %ld\n", ast_sched_when(sched, peer->expire)); ast_cli(fd, " Insecure : %s\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE))); - ast_cli(fd, " Force rport : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT))); + ast_cli(fd, " Force rport : %s\n", force_rport_string(peer->flags)); + ast_cli(fd, " Symmetric RTP: %s\n", comedia_string(peer->flags)); ast_cli(fd, " ACL : %s\n", AST_CLI_YESNO(peer->ha != NULL)); ast_cli(fd, " DirectMedACL : %s\n", AST_CLI_YESNO(peer->directmediaha != NULL)); ast_cli(fd, " T.38 support : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT))); @@ -18254,7 +18304,12 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct astman_append(s, "Callerid: %s\r\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "")); astman_append(s, "RegExpire: %ld seconds\r\n", ast_sched_when(sched, peer->expire)); astman_append(s, "SIP-AuthInsecure: %s\r\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE))); - astman_append(s, "SIP-Forcerport: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT)?"Y":"N")); + astman_append(s, "SIP-Forcerport: %s\r\n", ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT) ? + (ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? "A" : "a") : + (ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? "Y" : "N")); + astman_append(s, "SIP-Comedia: %s\r\n", ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA) ? + (ast_test_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP) ? "A" : "a") : + (ast_test_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP) ? "Y" : "N")); astman_append(s, "ACL: %s\r\n", (peer->ha?"Y":"N")); astman_append(s, "SIP-CanReinvite: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N")); astman_append(s, "SIP-DirectMedia: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N")); @@ -18866,7 +18921,7 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_ ast_cli(a->fd, " Context: %s\n", sip_cfg.default_context); ast_cli(a->fd, " Record on feature: %s\n", sip_cfg.default_record_on_feature); ast_cli(a->fd, " Record off feature: %s\n", sip_cfg.default_record_off_feature); - ast_cli(a->fd, " Force rport: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_NAT_FORCE_RPORT))); + ast_cli(a->fd, " Force rport: %s\n", force_rport_string(global_flags)); ast_cli(a->fd, " DTMF: %s\n", dtmfmode2str(ast_test_flag(&global_flags[0], SIP_DTMF))); ast_cli(a->fd, " Qualify: %d\n", default_qualify); ast_cli(a->fd, " Use ClientCode: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_USECLIENTCODE))); @@ -19237,7 +19292,7 @@ static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_a ast_cli(a->fd, " Theoretical Address: %s\n", ast_sockaddr_stringify(&cur->sa)); ast_cli(a->fd, " Received Address: %s\n", ast_sockaddr_stringify(&cur->recv)); ast_cli(a->fd, " SIP Transfer mode: %s\n", transfermode2str(cur->allowtransfer)); - ast_cli(a->fd, " Force rport: %s\n", AST_CLI_YESNO(ast_test_flag(&cur->flags[0], SIP_NAT_FORCE_RPORT))); + ast_cli(a->fd, " Force rport: %s\n", force_rport_string(cur->flags)); if (ast_sockaddr_isnull(&cur->redirip)) { ast_cli(a->fd, " Audio IP: %s (local)\n", @@ -27692,8 +27747,8 @@ static int handle_t38_options(struct ast_flags *flags, struct ast_flags *mask, s /*! \brief Handle flag-type options common to configuration of devices - peers - \param flags array of two struct ast_flags - \param mask array of two struct ast_flags + \param flags array of three struct ast_flags + \param mask array of three struct ast_flags \param v linked list of config variables to process \returns non-zero if any config options were handled, zero otherwise */ @@ -27743,19 +27798,7 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask ast_set_flag(&flags[0], SIP_DTMF_RFC2833); } } else if (!strcasecmp(v->name, "nat")) { - ast_set_flag(&mask[0], SIP_NAT_FORCE_RPORT); - ast_set_flag(&flags[0], SIP_NAT_FORCE_RPORT); /* Default to "force_rport" */ - if (!strcasecmp(v->value, "no")) { - ast_clear_flag(&flags[0], SIP_NAT_FORCE_RPORT); - } else if (!strcasecmp(v->value, "yes")) { - /* We've already defaulted to force_rport */ - ast_set_flag(&mask[1], SIP_PAGE2_SYMMETRICRTP); - ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP); - } else if (!strcasecmp(v->value, "comedia")) { - ast_clear_flag(&flags[0], SIP_NAT_FORCE_RPORT); - ast_set_flag(&mask[1], SIP_PAGE2_SYMMETRICRTP); - ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP); - } + sip_parse_nat_option(v->value, mask, flags); } else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) { ast_set_flag(&mask[0], SIP_REINVITE); ast_clear_flag(&flags[0], SIP_REINVITE); @@ -28969,7 +29012,7 @@ static int reload_config(enum channelreloadreason reason) struct sip_peer *peer; char *cat, *stringp, *context, *oldregcontext; char newcontexts[AST_MAX_CONTEXT], oldcontexts[AST_MAX_CONTEXT]; - struct ast_flags dummy[2]; + struct ast_flags dummy[3]; struct ast_flags config_flags = { reason == CHANNEL_MODULE_LOAD ? 0 : ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) ? 0 : CONFIG_FLAG_FILEUNCHANGED }; int auto_sip_domains = FALSE; struct ast_sockaddr old_bindaddr = bindaddr; @@ -29164,7 +29207,7 @@ static int reload_config(enum channelreloadreason reason) ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten)); ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */ ast_set_flag(&global_flags[0], SIP_DIRECT_MEDIA); /*!< Allow re-invites */ - ast_set_flag(&global_flags[0], SIP_NAT_FORCE_RPORT); /*!< Default to nat=force_rport */ + ast_set_flag(&global_flags[2], SIP_PAGE3_NAT_AUTO_RPORT); /*!< Default to nat=auto_force_rport */ ast_copy_string(default_engine, DEFAULT_ENGINE, sizeof(default_engine)); ast_copy_string(default_parkinglot, DEFAULT_PARKINGLOT, sizeof(default_parkinglot)); diff --git a/channels/sip/config_parser.c b/channels/sip/config_parser.c index 72a7b2c7d..55695e545 100644 --- a/channels/sip/config_parser.c +++ b/channels/sip/config_parser.c @@ -765,11 +765,128 @@ AST_TEST_DEFINE(sip_parse_host_line_test) } +/*! \brief Parse the comma-separated nat= option values */ +void sip_parse_nat_option(const char *value, struct ast_flags *mask, struct ast_flags *flags) +{ + char *parse, *this; + + if (!(parse = ast_strdupa(value))) { + return; + } + + /* Since we need to completely override the general settings if we are being called + * later for a peer, always set the flags for all options on the mask */ + ast_set_flag(&mask[0], SIP_NAT_FORCE_RPORT); + ast_set_flag(&mask[1], SIP_PAGE2_SYMMETRICRTP); + ast_set_flag(&mask[2], SIP_PAGE3_NAT_AUTO_RPORT); + ast_set_flag(&mask[2], SIP_PAGE3_NAT_AUTO_COMEDIA); + + while ((this = strsep(&parse, ","))) { + if (ast_false(this)) { + ast_clear_flag(&flags[0], SIP_NAT_FORCE_RPORT); + ast_clear_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP); + ast_clear_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT); + ast_clear_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA); + break; /* It doesn't make sense to have no + something else */ + } else if (!strcasecmp(this, "yes")) { + ast_log(LOG_WARNING, "nat=yes is deprecated, use nat=force_rport,comedia instead\n"); + ast_set_flag(&flags[0], SIP_NAT_FORCE_RPORT); + ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP); + ast_clear_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT); + ast_clear_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA); + break; /* It doesn't make sense to have yes + something else */ + } else if (!strcasecmp(this, "force_rport") && !ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) { + ast_set_flag(&flags[0], SIP_NAT_FORCE_RPORT); + } else if (!strcasecmp(this, "comedia") && !ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) { + ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP); + } else if (!strcasecmp(this, "auto_force_rport")) { + ast_set_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT); + /* In case someone did something dumb like nat=force_rport,auto_force_rport */ + ast_clear_flag(&flags[0], SIP_NAT_FORCE_RPORT); + } else if (!strcasecmp(this, "auto_comedia")) { + ast_set_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA); + /* In case someone did something dumb like nat=comedia,auto_comedia*/ + ast_clear_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP); + } + } +} + +#define TEST_FORCE_RPORT 1 << 0 +#define TEST_COMEDIA 1 << 1 +#define TEST_AUTO_FORCE_RPORT 1 << 2 +#define TEST_AUTO_COMEDIA 1 << 3 +static int match_nat_options(int val, struct ast_flags *flags) +{ + if ((!ast_test_flag(&flags[0], SIP_NAT_FORCE_RPORT)) != !(val & TEST_FORCE_RPORT)) { + return 0; + } + if (!ast_test_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP) != !(val & TEST_COMEDIA)) { + return 0; + } + if (!ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT) != !(val & TEST_AUTO_FORCE_RPORT)) { + return 0; + } + if (!ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA) != !(val & TEST_AUTO_COMEDIA)) { + return 0; + } + return 1; +} + +AST_TEST_DEFINE(sip_parse_nat_test) +{ + int i, res = AST_TEST_PASS; + struct ast_flags mask[3] = {{0}}, flags[3] = {{0}}; + struct { + const char *str; + int i; + } options[] = { + { "yes", TEST_FORCE_RPORT | TEST_COMEDIA }, + { "no", 0 }, + { "force_rport", TEST_FORCE_RPORT }, + { "comedia", TEST_COMEDIA }, + { "auto_force_rport", TEST_AUTO_FORCE_RPORT }, + { "auto_comedia", TEST_AUTO_COMEDIA }, + { "force_rport,auto_force_rport", TEST_AUTO_FORCE_RPORT }, + { "auto_force_rport,force_rport", TEST_AUTO_FORCE_RPORT }, + { "comedia,auto_comedia", TEST_AUTO_COMEDIA }, + { "auto_comedia,comedia", TEST_AUTO_COMEDIA }, + { "force_rport,comedia", TEST_FORCE_RPORT | TEST_COMEDIA }, + { "force_rport,auto_comedia", TEST_FORCE_RPORT | TEST_AUTO_COMEDIA }, + { "force_rport,yes,no", TEST_FORCE_RPORT | TEST_COMEDIA }, + { "auto_comedia,no,yes", 0 }, + }; + + switch (cmd) { + case TEST_INIT: + info->name = "sip_parse_nat_test"; + info->category = "/channels/chan_sip/"; + info->summary = "tests sip.conf nat line parsing"; + info->description = + "Tests parsing of various nat line configurations. " + "Verifies output matches expected behavior."; + return AST_TEST_NOT_RUN; + case TEST_EXECUTE: + break; + } + + for (i = 0; i < ARRAY_LEN(options); i++) { + sip_parse_nat_option(options[i].str, mask, flags); + if (!match_nat_options(options[i].i, flags)) { + ast_test_status_update(test, "Failed nat=%s\n", options[i].str); + res = AST_TEST_FAIL; + } + memset(flags, 0, sizeof(flags)); + memset(mask, 0, sizeof(mask)); + } + + return res; +} /*! \brief SIP test registration */ void sip_config_parser_register_tests(void) { AST_TEST_REGISTER(sip_parse_register_line_test); AST_TEST_REGISTER(sip_parse_host_line_test); + AST_TEST_REGISTER(sip_parse_nat_test); } /*! \brief SIP test registration */ @@ -777,5 +894,6 @@ void sip_config_parser_unregister_tests(void) { AST_TEST_UNREGISTER(sip_parse_register_line_test); AST_TEST_UNREGISTER(sip_parse_host_line_test); + AST_TEST_UNREGISTER(sip_parse_nat_test); } diff --git a/channels/sip/include/config_parser.h b/channels/sip/include/config_parser.h index 76fefc2c6..2737016e6 100644 --- a/channels/sip/include/config_parser.h +++ b/channels/sip/include/config_parser.h @@ -43,6 +43,18 @@ int sip_parse_register_line(struct sip_registry *reg, int default_expiry, const */ int sip_parse_host(char *line, int lineno, char **hostname, int *portnum, enum sip_transport *transport); +/*! \brief Parse the comma-separated nat= option values + * \param value The comma-separated value + * \param mask An array of ast_flags that will be set by this function + * and used as a mask for copying the flags later + * \param flags An array of ast_flags that will be set by this function + * + * \note The nat-related values in both mask and flags are assumed to empty. This function + * will treat the first "yes" or "no" value in a list of values as overiding all other values + * and will stop parsing. Auto values will override their non-auto counterparts. + */ +void sip_parse_nat_option(const char *value, struct ast_flags *mask, struct ast_flags *flags); + /*! * \brief register config parsing tests */ diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h index d5de40d9c..0ae8be8f8 100644 --- a/channels/sip/include/sip.h +++ b/channels/sip/include/sip.h @@ -363,9 +363,11 @@ #define SIP_PAGE3_SNOM_AOC (1 << 0) /*!< DPG: Allow snom aoc messages */ #define SIP_PAGE3_SRTP_TAG_32 (1 << 1) /*!< DP: Use a 32bit auth tag in INVITE not 80bit */ +#define SIP_PAGE3_NAT_AUTO_RPORT (1 << 2) /*!< DGP: Set SIP_NAT_FORCE_RPORT when NAT is detected */ +#define SIP_PAGE3_NAT_AUTO_COMEDIA (1 << 3) /*!< DGP: Set SIP_PAGE2_SYMMETRICRTP when NAT is detected */ #define SIP_PAGE3_FLAGS_TO_COPY \ - (SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32) + (SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32 | SIP_PAGE3_NAT_AUTO_RPORT | SIP_PAGE3_NAT_AUTO_COMEDIA) #define CHECK_AUTH_BUF_INITLEN 256 @@ -1062,6 +1064,7 @@ struct sip_pvt { * for incoming calls */ unsigned short req_secure_signaling:1;/*!< Whether we are required to have secure signaling or not */ + unsigned short natdetected:1; /*!< Whether we detected a NAT when processing the Via */ char tag[11]; /*!< Our tag for this session */ int timer_t1; /*!< SIP timer T1, ms rtt */ int timer_b; /*!< SIP timer B, ms */ diff --git a/channels/sip/include/sip_utils.h b/channels/sip/include/sip_utils.h index 47297ab68..01207e774 100644 --- a/channels/sip/include/sip_utils.h +++ b/channels/sip/include/sip_utils.h @@ -80,4 +80,10 @@ int hangup_sip2cause(int cause); */ const char *hangup_cause2sip(int cause); +/*! \brief Return a string describing the force_rport value for the given flags */ +const char *force_rport_string(struct ast_flags *flags); + +/*! \brief Return a string describing the comedia value for the given flags */ +const char *comedia_string(struct ast_flags *flags); + #endif diff --git a/channels/sip/utils.c b/channels/sip/utils.c new file mode 100644 index 000000000..14983ba82 --- /dev/null +++ b/channels/sip/utils.c @@ -0,0 +1,45 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 1999 - 2012, Digium, Inc. + * + * See http://www.asterisk.org for more information about + * the Asterisk project. Please do not directly contact + * any of the maintainers of this project for assistance; + * the project provides a web site, mailing lists and IRC + * channels for your use. + * + * This program is free software, distributed under the terms of + * the GNU General Public License Version 2. See the LICENSE file + * at the top of the source tree. + */ + +/*! + * \file + * \brief Utility functions for chan_sip + * + * \author Terry Wilson <twilson@digium.com> + */ + +#include "asterisk.h" + +#include "asterisk/utils.h" +#include "asterisk/cli.h" +#include "include/sip.h" +#include "include/sip_utils.h" + +const char *force_rport_string(struct ast_flags *flags) +{ + if (ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) { + return ast_test_flag(&flags[0], SIP_NAT_FORCE_RPORT) ? "Auto (Yes)" : "Auto (No)"; + } + return AST_CLI_YESNO(ast_test_flag(&flags[0], SIP_NAT_FORCE_RPORT)); +} + +const char *comedia_string(struct ast_flags *flags) +{ + if (ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) { + return ast_test_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP) ? "Auto (Yes)" : "Auto (No)"; + } + return AST_CLI_YESNO(ast_test_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP)); +} diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index a07f275a4..f64dc04a9 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -831,17 +831,34 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; However, this is only useful if the external traffic can reach us. ; The following settings are allowed (both globally and in individual sections): ; -; nat = no ; Use rport if the remote side says to use it. -; nat = force_rport ; Force rport to always be on. (default) -; nat = yes ; Force rport to always be on and perform comedia RTP handling. -; nat = comedia ; Use rport if the remote side says to use it and perform comedia RTP handling. +; nat = no ; Do no special NAT handling other than RFC3581 +; nat = force_rport ; Pretend there was an rport parameter even if there wasn't +; nat = comedia ; Send media to the port Asterisk received it from regardless +; ; of where the SDP says to send it. +; nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default) +; nat = auto_comedia ; Set the comedia option if Asterisk detects NAT +; +; The nat settings can be combined. For example, to set both force_rport and comedia +; one would set nat=force_rport,comedia. If any of the comma-separated options is 'no', +; Asterisk will ignore any other settings and set nat=no. If one of the "auto" settings +; is used in conjunction with its non-auto counterpart (nat=comedia,auto_comedia), then +; the non-auto option will be ignored. +; +; The RFC 3581-defined 'rport' parameter allows a client to request that Asterisk send +; SIP responses to it via the source IP and port from which the request originated +; instead of the address/port listed in the top-most Via header. This is useful if a +; client knows that it is behind a NAT and therefore cannot guess from what address/port +; its request will be sent. Asterisk will always honor the 'rport' parameter if it is +; sent. The force_rport setting causes Asterisk to always send responses back to the +; address/port from which it received requests; even if the other side doesn't support +; adding the 'rport' parameter. ; ; 'comedia RTP handling' refers to the technique of sending RTP to the port that the ; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is ; only partially related to RFC 4145 which was referred to as COMEDIA while it was in ; draft form. This method is used to accomodate endpoints that may be located behind -; NAT devices, and as such the port number they tell Asterisk to send RTP packets to -; for their media streams is not actual port number that will be used on the nearer +; NAT devices, and as such the address/port they tell Asterisk to send RTP packets to +; for their media streams is not the actual address/port that will be used on the nearer ; side of the NAT. ; ; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from |