merged jcmoore's patch for configurable SDP origin-field username and session field...

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83671 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d1b1547..db30792 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -553,6 +553,8 @@ static const struct cfsip_options {
 #define DEFAULT_MAX_CALL_BITRATE (384)         /*!< Max bitrate for video */
 #ifndef DEFAULT_USERAGENT
 #define DEFAULT_USERAGENT "Asterisk PBX"       /*!< Default Useragent: header unless re-defined in sip.conf */
+#define DEFAULT_SDPSESSION "Asterisk PBX"      /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
+#define DEFAULT_SDPOWNER "root"                /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
 #endif
 
 /* Default setttings are used as a channel setting and as a default when
@@ -614,6 +616,8 @@ static int dumphistory;                     /*!< Dump history to verbose before destroying SIP dia
 static char global_realm[MAXHOSTNAMELEN];              /*!< Default realm */
 static char global_regcontext[AST_MAX_CONTEXT];                /*!< Context for auto-extensions */
 static char global_useragent[AST_MAX_EXTENSION];       /*!< Useragent for the SIP channel */
+static char global_sdpsession[AST_MAX_EXTENSION];      /*!< SDP session name for the SIP channel */
+static char global_sdpowner[AST_MAX_EXTENSION];        /*!< SDP owner name for the SIP channel */
 static int allow_external_domains;     /*!< Accept calls to external SIP domains? */
 static int global_callevents;          /*!< Whether we send manager events or not */
 static int global_t1min;               /*!< T1 roundtrip time minimum */
@@ -6947,8 +6951,8 @@ static int add_t38_sdp(struct sip_request *resp, struct sip_pvt *p)
                        p->t38.jointcapability);
        }
        ast_str_append(&m_modem, 0, "v=0\r\n");
-       ast_str_append(&m_modem, 0, "o=root %d %d IN IP4 %s\r\n", p->sessionid, p->sessionversion, ast_inet_ntoa(udptldest.sin_addr));
-       ast_str_append(&m_modem, 0, "s=session\r\n");
+       ast_str_append(&m_modem, 0, "o=%s %d %d IN IP4 %s\r\n", ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner , p->sessionid, p->sessionversion, ast_inet_ntoa(udptldest.sin_addr));
+       ast_str_append(&m_modem, 0, "s=%s\r\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
        ast_str_append(&m_modem, 0, "c=IN IP4 %s\r\n", ast_inet_ntoa(udptldest.sin_addr));
        ast_str_append(&m_modem, 0, "t=0 0\r\n");
        ast_str_append(&m_modem, 0, "m=image %d udptl t38\r\n", ntohs(udptldest.sin_port));
@@ -7054,7 +7058,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
 
        /* SDP fields */
        char *version =         "v=0\r\n";              /* Protocol version */
-       char *subject =         "s=session\r\n";        /* Subject of the session */
+       char subject[256];                              /* Subject of the session */
        char owner[256];                                /* Session owner/creator */
        char connection[256];                           /* Connection data */
        char *stime = "t=0 0\r\n";                      /* Time the session is active */
@@ -7079,6 +7083,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
        char codecbuf[BUFSIZ];
        char buf[BUFSIZ];
 
+       /* Set the SDP session name */
+       snprintf(subject, sizeof(subject), "s=%s\r\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
+
        if (!p->rtp) {
                ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
                return AST_FAILURE;
@@ -7172,7 +7179,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
        /* We break with the "recommendation" and send our IP, in order that our
           peer doesn't have to ast_gethostbyname() us */
 
-       snprintf(owner, sizeof(owner), "o=root %d %d IN IP4 %s\r\n", p->sessionid, p->sessionversion, ast_inet_ntoa(dest.sin_addr));
+       snprintf(owner, sizeof(owner), "o=%s %d %d IN IP4 %s\r\n", ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner, p->sessionid, p->sessionversion, ast_inet_ntoa(dest.sin_addr));
        snprintf(connection, sizeof(connection), "c=IN IP4 %s\r\n", ast_inet_ntoa(dest.sin_addr));
        ast_str_append(&m_audio, 0, "m=audio %d RTP/AVP", ntohs(dest.sin_port));
 
@@ -11642,6 +11649,8 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
        ast_cli(a->fd, "  Call limit peers only:  %s\n", cli_yesno(global_limitonpeers));
        ast_cli(a->fd, "  Direct RTP setup:       %s\n", cli_yesno(global_directrtpsetup));
        ast_cli(a->fd, "  User Agent:             %s\n", global_useragent);
+       ast_cli(a->fd, "  SDP Session Name:       %s\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
+       ast_cli(a->fd, "  SDP Owner Name:         %s\n", ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner);
        ast_cli(a->fd, "  Reg. context:           %s\n", S_OR(global_regcontext, "(not set)"));
        ast_cli(a->fd, "  Regexten on Qualify:    %s\n", cli_yesno(global_regextenonqualify));
        ast_cli(a->fd, "  Caller ID:              %s\n", default_callerid);
@@ -17858,6 +17867,8 @@ static int reload_config(enum channelreloadreason reason)
        global_alwaysauthreject = 0;
        global_allowsubscribe = FALSE;
        snprintf(global_useragent, sizeof(global_useragent), "%s %s", DEFAULT_USERAGENT, ASTERISK_VERSION);
+       snprintf(global_sdpsession, sizeof(global_sdpsession), "%s %s", DEFAULT_SDPSESSION, ASTERISK_VERSION);
+       snprintf(global_sdpowner, sizeof(global_sdpowner), "%s", DEFAULT_SDPOWNER);
        ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime));
        ast_copy_string(global_realm, S_OR(ast_config_AST_SYSTEM_NAME, DEFAULT_REALM), sizeof(global_realm));
        ast_copy_string(default_callerid, DEFAULT_CALLERID, sizeof(default_callerid));
@@ -17930,6 +17941,14 @@ static int reload_config(enum channelreloadreason reason)
                } else if (!strcasecmp(v->name, "useragent")) {
                        ast_copy_string(global_useragent, v->value, sizeof(global_useragent));
                        ast_debug(1, "Setting SIP channel User-Agent Name to %s\n", global_useragent);
+               } else if (!strcasecmp(v->name, "sdpsession")) {
+                       ast_copy_string(global_sdpsession, v->value, sizeof(global_sdpsession));
+               } else if (!strcasecmp(v->name, "sdpowner")) {
+                       /* Field cannot contain spaces */
+                       if (!strstr(v->value, " "))
+                               ast_copy_string(global_sdpowner, v->value, sizeof(global_sdpowner));
+                       else
+                               ast_log(LOG_WARNING, "'%s' must not contain spaces at line %d.  Using default.\n", v->value, v->lineno);
                } else if (!strcasecmp(v->name, "allowtransfer")) {
                        global_allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED;
                } else if (!strcasecmp(v->name, "rtcachefriends")) {

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 6b96718..4675963 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -113,6 +113,11 @@ srvlookup=yes                      ; Enable DNS SRV lookups on outbound calls
                                ; The default user agent string also contains the Asterisk
                                ; version. If you don't want to expose this, change the
                                ; useragent string.
+;sdpsession=Asterisk PBX       ; Allows you to change the SDP session name string, (s=)
+                               ; Like the useragent parameter, the default user agent string
+                               ; also contains the Asterisk version.
+;sdpowner=root                 ; Allows you to change the username field in the SDP owner string, (o=)
+                               ; This field MUST NOT contain spaces
 ;promiscredir = no             ; If yes, allows 302 or REDIR to non-local SIP address
                                ; Note that promiscredir when redirects are made to the
                                        ; local system will cause loops since Asterisk is incapable