diff options
author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-01-22 20:56:09 +0000 |
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committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-01-22 20:56:09 +0000 |
commit | 2ca773694a0fa6dfe55abc1c3cd69a6977881057 (patch) | |
tree | 238fa43cb435783eb0bf7a099055aaa577ea1a0a /channels | |
parent | 35042556d86099c85858a523f2e95d59a22e9b25 (diff) |
Thanks to Russell's education I realize that BUFSIZ has changed since I learned the C language
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: http://svn.digium.com/svn/asterisk/branches/1.4@99652 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
-rw-r--r-- | channels/chan_sip.c | 84 |
1 files changed, 43 insertions, 41 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 1b7dfe33c..f09c72548 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -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, |