]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
Thanks to Russell's education I realize that BUFSIZ has changed since I learned the...
authorOlle Johansson <oej@edvina.net>
Tue, 22 Jan 2008 20:56:09 +0000 (20:56 +0000)
committerOlle Johansson <oej@edvina.net>
Tue, 22 Jan 2008 20:56:09 +0000 (20:56 +0000)
over 20 years ago... Resetting chan_sip to the size of BUFSIZ that I expected in my old
head to avoid to heavy memory allocations on some systems.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@99652 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 1b7dfe33c1ea75bd24ad6820801830c1cf9f6395..f09c725483d30a0a62455f7c79f9ae59fdb76a18 100644 (file)
@@ -159,6 +159,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #define TRUE     1
 #endif
 
+#define        SIPBUFSIZE              512
+
 #define XMIT_ERROR             -2
 
 #define VIDEO_CODEC_MASK        0x1fc0000 /*!< Video codecs from H.261 thru AST_FORMAT_MAX_VIDEO */
@@ -889,9 +891,9 @@ struct sip_refer {
        char referred_by[AST_MAX_EXTENSION];            /*!< Place to store REFERRED-BY extension */
        char referred_by_name[AST_MAX_EXTENSION];       /*!< Place to store REFERRED-BY extension */
        char refer_contact[AST_MAX_EXTENSION];          /*!< Place to store Contact info from a REFER extension */
-       char replaces_callid[BUFSIZ];                   /*!< Replace info: callid */
-       char replaces_callid_totag[BUFSIZ/2];           /*!< Replace info: to-tag */
-       char replaces_callid_fromtag[BUFSIZ/2];         /*!< Replace info: from-tag */
+       char replaces_callid[SIPBUFSIZE];                       /*!< Replace info: callid */
+       char replaces_callid_totag[SIPBUFSIZE/2];               /*!< Replace info: to-tag */
+       char replaces_callid_fromtag[SIPBUFSIZE/2];             /*!< Replace info: from-tag */
        struct sip_pvt *refer_call;                     /*!< Call we are referring */
        int attendedtransfer;                           /*!< Attended or blind transfer? */
        int localtransfer;                              /*!< Transfer to local domain? */
@@ -2984,7 +2986,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
        ast_set_flag(&p->flags[0], SIP_OUTGOING);
 
        if (p->options->transfer) {
-               char buf[BUFSIZ/2];
+               char buf[SIPBUFSIZE/2];
 
                if (referer) {
                        if (sipdebug && option_debug > 2)
@@ -3996,13 +3998,13 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
        /* Set the native formats for audio  and merge in video */
        tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | video;
        if (option_debug > 2) {
-               char buf[BUFSIZ];
-               ast_log(LOG_DEBUG, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, tmp->nativeformats));
-               ast_log(LOG_DEBUG, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->jointcapability));
-               ast_log(LOG_DEBUG, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->capability));
-               ast_log(LOG_DEBUG, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, ast_codec_choose(&i->prefs, what, 1)));
+               char buf[SIPBUFSIZE];
+               ast_log(LOG_DEBUG, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, tmp->nativeformats));
+               ast_log(LOG_DEBUG, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->jointcapability));
+               ast_log(LOG_DEBUG, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->capability));
+               ast_log(LOG_DEBUG, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, ast_codec_choose(&i->prefs, what, 1)));
                if (i->prefcodec)
-                       ast_log(LOG_DEBUG, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->prefcodec));
+                       ast_log(LOG_DEBUG, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->prefcodec));
        }
 
        /* XXX Why are we choosing a codec from the native formats?? */
@@ -5409,18 +5411,18 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
                
        if (debug) {
                /* shame on whoever coded this.... */
-               char s1[BUFSIZ], s2[BUFSIZ], s3[BUFSIZ], s4[BUFSIZ];
+               char s1[SIPBUFSIZE], s2[SIPBUFSIZE], s3[SIPBUFSIZE], s4[SIPBUFSIZE];
 
                ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s, combined - %s\n",
-                           ast_getformatname_multiple(s1, BUFSIZ, p->capability),
-                           ast_getformatname_multiple(s2, BUFSIZ, newpeercapability),
-                           ast_getformatname_multiple(s3, BUFSIZ, vpeercapability),
-                           ast_getformatname_multiple(s4, BUFSIZ, newjointcapability));
+                           ast_getformatname_multiple(s1, SIPBUFSIZE, p->capability),
+                           ast_getformatname_multiple(s2, SIPBUFSIZE, newpeercapability),
+                           ast_getformatname_multiple(s3, SIPBUFSIZE, vpeercapability),
+                           ast_getformatname_multiple(s4, SIPBUFSIZE, newjointcapability));
 
                ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
-                           ast_rtp_lookup_mime_multiple(s1, BUFSIZ, p->noncodeccapability, 0, 0),
-                           ast_rtp_lookup_mime_multiple(s2, BUFSIZ, peernoncodeccapability, 0, 0),
-                           ast_rtp_lookup_mime_multiple(s3, BUFSIZ, newnoncodeccapability, 0, 0));
+                           ast_rtp_lookup_mime_multiple(s1, SIPBUFSIZE, p->noncodeccapability, 0, 0),
+                           ast_rtp_lookup_mime_multiple(s2, SIPBUFSIZE, peernoncodeccapability, 0, 0),
+                           ast_rtp_lookup_mime_multiple(s3, SIPBUFSIZE, newnoncodeccapability, 0, 0));
        }
        if (!newjointcapability) {
                /* If T.38 was not negotiated either, totally bail out... */
@@ -5474,8 +5476,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 
        /* Ok, we're going with this offer */
        if (option_debug > 1) {
-               char buf[BUFSIZ];
-               ast_log(LOG_DEBUG, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, BUFSIZ, p->jointcapability));
+               char buf[SIPBUFSIZE];
+               ast_log(LOG_DEBUG, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, p->jointcapability));
        }
 
        if (!p->owner)  /* There's no open channel owning us so we can return here. For a re-invite or so, we proceed */
@@ -5486,10 +5488,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 
        if (!(p->owner->nativeformats & p->jointcapability) && (p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
                if (debug) {
-                       char s1[BUFSIZ], s2[BUFSIZ];
+                       char s1[SIPBUFSIZE], s2[SIPBUFSIZE];
                        ast_log(LOG_DEBUG, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n", 
-                               ast_getformatname_multiple(s1, BUFSIZ, p->jointcapability),
-                               ast_getformatname_multiple(s2, BUFSIZ, p->owner->nativeformats));
+                               ast_getformatname_multiple(s1, SIPBUFSIZE, p->jointcapability),
+                               ast_getformatname_multiple(s2, SIPBUFSIZE, p->owner->nativeformats));
                }
                p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability);
                ast_set_read_format(p->owner, p->owner->readformat);
@@ -5685,7 +5687,7 @@ static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, const st
 /*! \brief Add route header into request per learned route */
 static void add_route(struct sip_request *req, struct sip_route *route)
 {
-       char r[BUFSIZ*2], *p;
+       char r[SIPBUFSIZE*2], *p;
        int n, rem = sizeof(r);
 
        if (!route)
@@ -5839,7 +5841,7 @@ static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg
                snprintf(tmp, sizeof(tmp), "%d", p->expiry);
                add_header(resp, "Expires", tmp);
                if (p->expiry) {        /* Only add contact if we have an expiry time */
-                       char contact[BUFSIZ];
+                       char contact[SIPBUFSIZE];
                        snprintf(contact, sizeof(contact), "%s;expires=%d", p->our_contact, p->expiry);
                        add_header(resp, "Contact", contact);   /* Not when we unregister */
                }
@@ -6427,7 +6429,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
 
 
        if (option_debug > 1) {
-               char codecbuf[BUFSIZ];
+               char codecbuf[SIPBUFSIZE];
                ast_log(LOG_DEBUG, "** Our capability: %s Video flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability), ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False");
                ast_log(LOG_DEBUG, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
        }
@@ -6604,8 +6606,8 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
        p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
 
        if (option_debug > 2) {
-               char buf[BUFSIZ];
-               ast_log(LOG_DEBUG, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, BUFSIZ, capability));
+               char buf[SIPBUFSIZE];
+               ast_log(LOG_DEBUG, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, capability));
        }
 
        return AST_SUCCESS;
@@ -6769,7 +6771,7 @@ static int transmit_reinvite_with_t38_sdp(struct sip_pvt *p)
 /*! \brief Check Contact: URI of SIP message */
 static void extract_uri(struct sip_pvt *p, struct sip_request *req)
 {
-       char stripped[BUFSIZ];
+       char stripped[SIPBUFSIZE];
        char *c;
 
        ast_copy_string(stripped, get_header(req, "Contact"), sizeof(stripped));
@@ -6876,8 +6878,8 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
        size_t invite_max = sizeof(invite_buf);
        char from[256];
        char to[256];
-       char tmp[BUFSIZ/2];
-       char tmp2[BUFSIZ/2];
+       char tmp[SIPBUFSIZE/2];
+       char tmp2[SIPBUFSIZE/2];
        const char *l = NULL, *n = NULL;
        const char *urioptions = "";
 
@@ -7024,7 +7026,7 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
        append_date(&req);
        if (sipmethod == SIP_REFER) {   /* Call transfer */
                if (p->refer) {
-                       char buf[BUFSIZ];
+                       char buf[SIPBUFSIZE];
                        if (!ast_strlen_zero(p->refer->refer_to))
                                add_header(&req, "Refer-To", p->refer->refer_to);
                        if (!ast_strlen_zero(p->refer->referred_by)) {
@@ -7337,7 +7339,7 @@ static int transmit_sip_request(struct sip_pvt *p, struct sip_request *req)
 static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate)
 {
        struct sip_request req;
-       char tmp[BUFSIZ/2];
+       char tmp[SIPBUFSIZE/2];
 
        reqprep(&req, p, SIP_NOTIFY, 0, 1);
        snprintf(tmp, sizeof(tmp), "refer;id=%d", cseq);
@@ -7924,7 +7926,7 @@ static void reg_source_db(struct sip_peer *peer)
 /*! \brief Save contact header for 200 OK on INVITE */
 static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req)
 {
-       char contact[BUFSIZ]; 
+       char contact[SIPBUFSIZE]; 
        char *c;
 
        /* Look for brackets */
@@ -8008,8 +8010,8 @@ static int set_address_from_contact(struct sip_pvt *pvt)
 /*! \brief Parse contact header and save registration (peer registration) */
 static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
 {
-       char contact[BUFSIZ]; 
-       char data[BUFSIZ];
+       char contact[SIPBUFSIZE]; 
+       char data[SIPBUFSIZE];
        const char *expires = get_header(req, "Expires");
        int expiry = atoi(expires);
        char *curi, *n, *pt;
@@ -10581,7 +10583,7 @@ static int sip_show_settings(int fd, int argc, char *argv[])
 {
        int realtimepeers;
        int realtimeusers;
-       char codec_buf[BUFSIZ];
+       char codec_buf[SIPBUFSIZE];
 
        realtimepeers = ast_check_realtime("sippeers");
        realtimeusers = ast_check_realtime("sipusers");
@@ -10745,7 +10747,7 @@ static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions
                        referstatus = referstatus2str(cur->refer->status);
                }
                if (cur->subscribed == NONE && !subscriptions) {
-                       char formatbuf[BUFSIZ/2];
+                       char formatbuf[SIPBUFSIZE/2];
                        ast_cli(fd, FORMAT, ast_inet_ntoa(cur->sa.sin_addr), 
                                S_OR(cur->username, S_OR(cur->cid_num, "(None)")),
                                cur->callid, 
@@ -10925,7 +10927,7 @@ static int sip_show_channel(int fd, int argc, char *argv[])
        ast_mutex_lock(&iflock);
        for (cur = iflist; cur; cur = cur->next) {
                if (!strncasecmp(cur->callid, argv[3], len)) {
-                       char formatbuf[BUFSIZ/2];
+                       char formatbuf[SIPBUFSIZE/2];
                        ast_cli(fd,"\n");
                        if (cur->subscribed != NONE)
                                ast_cli(fd, "  * Subscription (type: %s)\n", subscription_type2str(cur->subscribed));
@@ -11881,7 +11883,7 @@ static struct ast_custom_function sipchaninfo_function = {
 /*! \brief Parse 302 Moved temporalily response */
 static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req)
 {
-       char tmp[BUFSIZ];
+       char tmp[SIPBUFSIZE];
        char *s, *e, *uri, *t;
        char *domain;
 
@@ -14482,7 +14484,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
        }
        /* Generate a Replaces string to be used in the INVITE during attended transfer */
        if (p->refer->replaces_callid && !ast_strlen_zero(p->refer->replaces_callid)) {
-               char tempheader[BUFSIZ];
+               char tempheader[SIPBUFSIZE];
                snprintf(tempheader, sizeof(tempheader), "%s%s%s%s%s", p->refer->replaces_callid, 
                                p->refer->replaces_callid_totag ? ";to-tag=" : "", 
                                p->refer->replaces_callid_totag,