diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 674c14452..13e21737d 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -1401,10 +1401,10 @@ static const char *get_sdp(struct sip_request *req, const char *name); static int find_sdp(struct sip_request *req); static int process_sdp(struct sip_pvt *p, struct sip_request *req); static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate, - char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, + struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size); static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate, - char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, + struct ast_str **m_buf, struct ast_str **a_buf, int debug); static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p); static void do_setnat(struct sip_pvt *p, int natflags); @@ -6503,7 +6503,7 @@ static int add_vidupdate(struct sip_request *req) /*! \brief Add codec offer to SDP offer/answer body in INVITE or 200 OK */ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate, - char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, + struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size) { int rtp_code; @@ -6520,20 +6520,20 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate fmt = ast_codec_pref_getsize(pref, codec); } else /* I dont see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */ return; - ast_build_string(m_buf, m_size, " %d", rtp_code); - ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code, + ast_str_append(m_buf, 0, " %d", rtp_code); + ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, ast_rtp_lookup_mime_subtype(1, codec, ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0), sample_rate); if (codec == AST_FORMAT_G729A) { /* Indicate that we don't support VAD (G.729 annex B) */ - ast_build_string(a_buf, a_size, "a=fmtp:%d annexb=no\r\n", rtp_code); + ast_str_append(a_buf, 0, "a=fmtp:%d annexb=no\r\n", rtp_code); } else if (codec == AST_FORMAT_G723_1) { /* Indicate that we don't support VAD (G.723.1 annex A) */ - ast_build_string(a_buf, a_size, "a=fmtp:%d annexa=no\r\n", rtp_code); + ast_str_append(a_buf, 0, "a=fmtp:%d annexa=no\r\n", rtp_code); } else if (codec == AST_FORMAT_ILBC) { /* Add information about us using only 20/30 ms packetization */ - ast_build_string(a_buf, a_size, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms); + ast_str_append(a_buf, 0, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms); } if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size)) @@ -6547,7 +6547,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate /*! \brief Add video codec offer to SDP offer/answer body in INVITE or 200 OK */ /* This is different to the audio one now so we can add more caps later */ static void add_vcodec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate, - char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, + struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size) { int rtp_code; @@ -6561,15 +6561,15 @@ static void add_vcodec_to_sdp(const struct sip_pvt *p, int codec, int sample_rat if ((rtp_code = ast_rtp_lookup_code(p->vrtp, 1, codec)) == -1) return; - ast_build_string(m_buf, m_size, " %d", rtp_code); - ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code, + ast_str_append(m_buf, 0, " %d", rtp_code); + ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, ast_rtp_lookup_mime_subtype(1, codec, 0), sample_rate); /* Add fmtp code here */ } /*! \brief Add text codec offer to SDP offer/answer body in INVITE or 200 OK */ static void add_tcodec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate, - char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, + struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size) { int rtp_code; @@ -6583,8 +6583,8 @@ static void add_tcodec_to_sdp(const struct sip_pvt *p, int codec, int sample_rat if ((rtp_code = ast_rtp_lookup_code(p->trtp, 1, codec)) == -1) return; - ast_build_string(m_buf, m_size, " %d", rtp_code); - ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code, + ast_str_append(m_buf, 0, " %d", rtp_code); + ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, ast_rtp_lookup_mime_subtype(1, codec, 0), sample_rate); /* Add fmtp code here */ } @@ -6625,17 +6625,8 @@ static int add_t38_sdp(struct sip_request *resp, struct sip_pvt *p) int len = 0; int x = 0; struct sockaddr_in udptlsin; - char v[256] = ""; - char s[256] = ""; - char o[256] = ""; - char c[256] = ""; - char t[256] = ""; - char m_modem[256]; - char a_modem[1024]; - char *m_modem_next = m_modem; - size_t m_modem_left = sizeof(m_modem); - char *a_modem_next = a_modem; - size_t a_modem_left = sizeof(a_modem); + struct ast_str *m_modem = ast_str_alloca(1024); + struct ast_str *a_modem = ast_str_alloca(1024); struct sockaddr_in udptldest = { 0, }; int debug; @@ -6676,39 +6667,34 @@ static int add_t38_sdp(struct sip_request *resp, struct sip_pvt *p) p->t38.peercapability, p->t38.jointcapability); } - snprintf(v, sizeof(v), "v=0\r\n"); - snprintf(o, sizeof(o), "o=root %d %d IN IP4 %s\r\n", p->sessionid, p->sessionversion, ast_inet_ntoa(udptldest.sin_addr)); - snprintf(s, sizeof(s), "s=session\r\n"); - snprintf(c, sizeof(c), "c=IN IP4 %s\r\n", ast_inet_ntoa(udptldest.sin_addr)); - snprintf(t, sizeof(t), "t=0 0\r\n"); - ast_build_string(&m_modem_next, &m_modem_left, "m=image %d udptl t38\r\n", ntohs(udptldest.sin_port)); + 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, "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)); if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_0) - ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxVersion:0\r\n"); + ast_str_append(&a_modem, 0, "a=T38FaxVersion:0\r\n"); if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_1) - ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxVersion:1\r\n"); + ast_str_append(&a_modem, 0, "a=T38FaxVersion:1\r\n"); if ((x = t38_get_rate(p->t38.jointcapability))) - ast_build_string(&a_modem_next, &a_modem_left, "a=T38MaxBitRate:%d\r\n",x); - ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxFillBitRemoval:%d\r\n", (p->t38.jointcapability & T38FAX_FILL_BIT_REMOVAL) ? 1 : 0); - ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxTranscodingMMR:%d\r\n", (p->t38.jointcapability & T38FAX_TRANSCODING_MMR) ? 1 : 0); - ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxTranscodingJBIG:%d\r\n", (p->t38.jointcapability & T38FAX_TRANSCODING_JBIG) ? 1 : 0); - ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxRateManagement:%s\r\n", (p->t38.jointcapability & T38FAX_RATE_MANAGEMENT_LOCAL_TCF) ? "localTCF" : "transferredTCF"); + ast_str_append(&a_modem, 0, "a=T38MaxBitRate:%d\r\n",x); + ast_str_append(&a_modem, 0, "a=T38FaxFillBitRemoval:%d\r\n", (p->t38.jointcapability & T38FAX_FILL_BIT_REMOVAL) ? 1 : 0); + ast_str_append(&a_modem, 0, "a=T38FaxTranscodingMMR:%d\r\n", (p->t38.jointcapability & T38FAX_TRANSCODING_MMR) ? 1 : 0); + ast_str_append(&a_modem, 0, "a=T38FaxTranscodingJBIG:%d\r\n", (p->t38.jointcapability & T38FAX_TRANSCODING_JBIG) ? 1 : 0); + ast_str_append(&a_modem, 0, "a=T38FaxRateManagement:%s\r\n", (p->t38.jointcapability & T38FAX_RATE_MANAGEMENT_LOCAL_TCF) ? "localTCF" : "transferredTCF"); x = ast_udptl_get_local_max_datagram(p->udptl); - ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxMaxBuffer:%d\r\n",x); - ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxMaxDatagram:%d\r\n",x); + ast_str_append(&a_modem, 0, "a=T38FaxMaxBuffer:%d\r\n",x); + ast_str_append(&a_modem, 0, "a=T38FaxMaxDatagram:%d\r\n",x); if (p->t38.jointcapability != T38FAX_UDP_EC_NONE) - ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxUdpEC:%s\r\n", (p->t38.jointcapability & T38FAX_UDP_EC_REDUNDANCY) ? "t38UDPRedundancy" : "t38UDPFEC"); - len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + strlen(m_modem) + strlen(a_modem); + ast_str_append(&a_modem, 0, "a=T38FaxUdpEC:%s\r\n", (p->t38.jointcapability & T38FAX_UDP_EC_REDUNDANCY) ? "t38UDPRedundancy" : "t38UDPFEC"); + len = m_modem->used + a_modem->used; add_header(resp, "Content-Type", "application/sdp"); add_header_contentLength(resp, len); - add_line(resp, v); - add_line(resp, o); - add_line(resp, s); - add_line(resp, c); - add_line(resp, t); - add_line(resp, m_modem); - add_line(resp, a_modem); - + add_line(resp, m_modem->str); + add_line(resp, a_modem->str); + /* Update lastrtprx when we send our SDP */ p->lastrtprx = p->lastrtptx = time(NULL); @@ -6718,7 +6704,7 @@ static int add_t38_sdp(struct sip_request *resp, struct sip_pvt *p) /*! \brief Add RFC 2833 DTMF offer to SDP */ static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate, - char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, + struct ast_str **m_buf, struct ast_str **a_buf, int debug) { int rtp_code; @@ -6728,19 +6714,20 @@ static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_ if ((rtp_code = ast_rtp_lookup_code(p->rtp, 0, format)) == -1) return; - ast_build_string(m_buf, m_size, " %d", rtp_code); - ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code, + ast_str_append(m_buf, 0, " %d", rtp_code); + ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, ast_rtp_lookup_mime_subtype(0, format, 0), sample_rate); - if (format == AST_RTP_DTMF) - /* Indicate we support DTMF and FLASH... */ - ast_build_string(a_buf, a_size, "a=fmtp:%d 0-16\r\n", rtp_code); + if (format == AST_RTP_DTMF) /* Indicate we support DTMF and FLASH... */ + ast_str_append(a_buf, 0, "a=fmtp:%d 0-16\r\n", rtp_code); } /*! \brief Set all IP media addresses for this call \note called from add_sdp() */ -static void get_our_media_address(struct sip_pvt *p, int needvideo, struct sockaddr_in *sin, struct sockaddr_in *vsin, struct sockaddr_in *tsin, struct sockaddr_in *dest, struct sockaddr_in *vdest) +static void get_our_media_address(struct sip_pvt *p, int needvideo, + struct sockaddr_in *sin, struct sockaddr_in *vsin, struct sockaddr_in *tsin, + struct sockaddr_in *dest, struct sockaddr_in *vdest) { /* First, get our address */ ast_rtp_get_us(p->rtp, sin); @@ -6794,24 +6781,12 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) char *stime = "t=0 0\r\n"; /* Time the session is active */ char bandwidth[256] = ""; /* Max bitrate */ char *hold; - char m_audio[256]; /* Media declaration line for audio */ - char m_video[256]; /* Media declaration line for video */ - char m_text[256]; /* Media declaration line for text */ - char a_audio[1024]; /* Attributes for audio */ - char a_video[1024]; /* Attributes for video */ - char a_text[1024]; /* Attributes for text */ - char *m_audio_next = m_audio; - char *m_video_next = m_video; - char *m_text_next = m_text; - size_t m_audio_left = sizeof(m_audio); - size_t m_video_left = sizeof(m_video); - size_t m_text_left = sizeof(m_text); - char *a_audio_next = a_audio; - char *a_video_next = a_video; - char *a_text_next = a_text; - size_t a_audio_left = sizeof(a_audio); - size_t a_video_left = sizeof(a_video); - size_t a_text_left = sizeof(a_text); + struct ast_str *m_audio = ast_str_alloca(256); /* Media declaration line for audio */ + struct ast_str *m_video = ast_str_alloca(256); /* Media declaration line for video */ + struct ast_str *m_text = ast_str_alloca(256); /* Media declaration line for text */ + struct ast_str *a_audio = ast_str_alloca(1024); /* Attributes for audio */ + struct ast_str *a_video = ast_str_alloca(1024); /* Attributes for video */ + struct ast_str *a_text = ast_str_alloca(1024); /* Attributes for text */ int x; int capability; @@ -6825,9 +6800,6 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) char codecbuf[BUFSIZ]; char buf[BUFSIZ]; - m_video[0] = '\0'; /* Reset the video media string if it's not needed */ - m_text[0] = '\0'; /* Reset the video media string if it's not needed */ - if (!p->rtp) { ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n"); return AST_FAILURE; @@ -6850,8 +6822,8 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) #ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_RTP)) { - ast_build_string(&m_audio_next, &m_audio_left, " %d", 191); - ast_build_string(&a_audio_next, &a_audio_left, "a=rtpmap:%d %s/%d\r\n", 191, "t38", 8000); + ast_str_append(&m_audio, 0, " %d", 191); + ast_str_append(&a_audio, 0, "a=rtpmap:%d %s/%d\r\n", 191, "t38", 8000); } #endif @@ -6873,7 +6845,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) /* 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) { - ast_build_string(&m_video_next, &m_video_left, "m=video %d RTP/AVP", ntohs(vdest.sin_port)); + ast_str_append(&m_video, 0, "m=video %d RTP/AVP", ntohs(vdest.sin_port)); /* Build max bitrate string */ if (p->maxcallbitrate) @@ -6884,9 +6856,11 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) /* Check if we need text in this call */ if((capability & AST_FORMAT_TEXT_MASK) && !ast_test_flag(&p->flags[1], SIP_PAGE2_NOTEXT)) { - if (sipdebug_text) ast_verbose("We think we can do text\n"); + if (sipdebug_text) + ast_verbose("We think we can do text\n"); if (p->trtp) { - if (sipdebug_text) ast_verbose("And we have a text rtp object\n"); + if (sipdebug_text) + ast_verbose("And we have a text rtp object\n"); needtext = TRUE; ast_debug(2, "This call needs text offers! \n"); } else @@ -6896,7 +6870,8 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) /* Ok, we need text. Let's add what we need for text and set codecs. Text is handled differently than audio since we can not transcode. */ if (needtext) { - if (sipdebug_text) ast_verbose("Lets set up the text sdp\n"); + if (sipdebug_text) + ast_verbose("Lets set up the text sdp\n"); /* Determine text destination */ if (p->tredirip.sin_addr.s_addr) { tdest.sin_addr = p->tredirip.sin_addr; @@ -6905,7 +6880,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) tdest.sin_addr = p->ourip.sin_addr; tdest.sin_port = tsin.sin_port; } - ast_build_string(&m_text_next, &m_text_left, "m=text %d RTP/AVP", ntohs(tdest.sin_port)); + ast_str_append(&m_text, 0, "m=text %d RTP/AVP", ntohs(tdest.sin_port)); if (debug) /* XXX should I use tdest below ? */ ast_verbose("Text is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(tsin.sin_port)); @@ -6919,7 +6894,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) snprintf(owner, sizeof(owner), "o=root %d %d IN IP4 %s\r\n", 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_build_string(&m_audio_next, &m_audio_left, "m=audio %d RTP/AVP", ntohs(dest.sin_port)); + ast_str_append(&m_audio, 0, "m=audio %d RTP/AVP", ntohs(dest.sin_port)); if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_ONEDIR) hold = "a=recvonly\r\n"; @@ -6941,8 +6916,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) int codec = p->prefcodec & AST_FORMAT_AUDIO_MASK; add_codec_to_sdp(p, codec, SDP_SAMPLE_RATE(codec), - &m_audio_next, &m_audio_left, - &a_audio_next, &a_audio_left, + &m_audio, &a_audio, debug, &min_audio_packet_size); alreadysent |= codec; } @@ -6961,8 +6935,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) continue; add_codec_to_sdp(p, codec, SDP_SAMPLE_RATE(codec), - &m_audio_next, &m_audio_left, - &a_audio_next, &a_audio_left, + &m_audio, &a_audio, debug, &min_audio_packet_size); alreadysent |= codec; } @@ -6977,19 +6950,13 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) if (x <= AST_FORMAT_MAX_AUDIO) add_codec_to_sdp(p, x, SDP_SAMPLE_RATE(x), - &m_audio_next, &m_audio_left, - &a_audio_next, &a_audio_left, - debug, &min_audio_packet_size); + &m_audio, &a_audio, debug, &min_audio_packet_size); else if (x <= AST_FORMAT_MAX_VIDEO) add_vcodec_to_sdp(p, x, 90000, - &m_video_next, &m_video_left, - &a_video_next, &a_video_left, - debug, &min_video_packet_size); + &m_video, &a_video, debug, &min_video_packet_size); else if (x <= AST_FORMAT_MAX_TEXT) add_tcodec_to_sdp(p, x, 1000, - &m_text_next, &m_text_left, - &a_text_next, &a_text_left, - debug, &min_text_packet_size); + &m_text, &a_text, debug, &min_text_packet_size); } /* Now add DTMF RFC2833 telephony-event as a codec */ @@ -6997,43 +6964,42 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) if (!(p->jointnoncodeccapability & x)) continue; - add_noncodec_to_sdp(p, x, 8000, - &m_audio_next, &m_audio_left, - &a_audio_next, &a_audio_left, - debug); + add_noncodec_to_sdp(p, x, 8000, &m_audio, &a_audio, debug); } ast_debug(3, "-- Done with adding codecs to SDP\n"); if (!p->owner || !ast_internal_timing_enabled(p->owner)) - ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - -\r\n"); + ast_str_append(&a_audio, 0, "a=silenceSupp:off - - - -\r\n"); if (min_audio_packet_size) - ast_build_string(&a_audio_next, &a_audio_left, "a=ptime:%d\r\n", min_audio_packet_size); + ast_str_append(&a_audio, 0, "a=ptime:%d\r\n", min_audio_packet_size); /* XXX don't think you can have ptime for video */ if (min_video_packet_size) - ast_build_string(&a_video_next, &a_video_left, "a=ptime:%d\r\n", min_video_packet_size); + ast_str_append(&a_video, 0, "a=ptime:%d\r\n", min_video_packet_size); /* XXX don't think you can have ptime for text */ if (min_text_packet_size) - ast_build_string(&a_text_next, &a_text_left, "a=ptime:%d\r\n", min_text_packet_size); + ast_str_append(&a_text, 0, "a=ptime:%d\r\n", min_text_packet_size); - if ((m_audio_left < 2) || (m_video_left < 2) || (m_text_left < 2) || - (a_audio_left == 0) || (a_video_left == 0) || (a_text_left == 0)) + if (m_audio->len - m_audio->used < 2 || m_video->len - m_video->used < 2 || + m_text->len - m_text->used < 2 || a_text->len - a_text->used < 2 || + a_audio->len - a_audio->used < 2 || a_video->len - a_video->used < 2) ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n"); - ast_build_string(&m_audio_next, &m_audio_left, "\r\n"); + ast_str_append(&m_audio, 0, "\r\n"); if (needvideo) - ast_build_string(&m_video_next, &m_video_left, "\r\n"); + ast_str_append(&m_video, 0, "\r\n"); if (needtext) - ast_build_string(&m_text_next, &m_text_left, "\r\n"); + ast_str_append(&m_text, 0, "\r\n"); - len = strlen(version) + strlen(subject) + strlen(owner) + strlen(connection) + strlen(stime) + strlen(m_audio) + strlen(a_audio) + strlen(hold); + len = strlen(version) + strlen(subject) + strlen(owner) + + strlen(connection) + strlen(stime) + m_audio->used + a_audio->used + strlen(hold); if (needvideo) /* only if video response is appropriate */ - len += strlen(m_video) + strlen(a_video) + strlen(bandwidth) + strlen(hold); + len += m_video->used + a_video->used + strlen(bandwidth) + strlen(hold); if (needtext) /* only if text response is appropriate */ - len += strlen(m_text) + strlen(a_text) + strlen(hold); + len += m_text->used + a_text->used + strlen(hold); add_header(resp, "Content-Type", "application/sdp"); add_header_contentLength(resp, len); @@ -7044,17 +7010,17 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) if (needvideo) /* only if video response is appropriate */ add_line(resp, bandwidth); add_line(resp, stime); - add_line(resp, m_audio); - add_line(resp, a_audio); + add_line(resp, m_audio->str); + add_line(resp, a_audio->str); add_line(resp, hold); if (needvideo) { /* only if video response is appropriate */ - add_line(resp, m_video); - add_line(resp, a_video); + add_line(resp, m_video->str); + add_line(resp, a_video->str); add_line(resp, hold); /* Repeat hold for the video stream */ } if (needtext) { /* only if text response is appropriate */ - add_line(resp, m_text); - add_line(resp, a_text); + add_line(resp, m_text->str); + add_line(resp, a_text->str); add_line(resp, hold); /* Repeat hold for the text stream */ } @@ -7323,14 +7289,13 @@ static void build_rpid(struct sip_pvt *p) /*! \brief Initiate new SIP request to peer/user */ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod) { - char invite_buf[256] = ""; - char *invite = invite_buf; - size_t invite_max = sizeof(invite_buf); + struct ast_str *invite = ast_str_alloca(256); char from[256]; char to[256]; - char tmp[BUFSIZ/2]; - char tmp2[BUFSIZ/2]; - const char *l = NULL, *n = NULL; + char tmp_n[BUFSIZ/2]; /* build a local copy of 'n' if needed */ + char tmp_l[BUFSIZ/2]; /* build a local copy of 'l' if needed */ + const char *l = NULL; /* XXX what is this, exactly ? */ + const char *n = NULL; /* XXX what is this, exactly ? */ const char *urioptions = ""; if (ast_test_flag(&p->flags[0], SIP_USEREQPHONE)) { @@ -7382,10 +7347,10 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho ast_string_field_set(p, fromname, n); if (pedanticsipchecking) { - ast_uri_encode(n, tmp, sizeof(tmp), 0); - n = tmp; - ast_uri_encode(l, tmp2, sizeof(tmp2), 0); - l = tmp2; + ast_uri_encode(n, tmp_n, sizeof(tmp_n), 0); + n = tmp_n; + ast_uri_encode(l, tmp_l, sizeof(tmp_l), 0); + l = tmp_l; } if (ntohs(p->ourip.sin_port) != STANDARD_SIP_PORT && ast_strlen_zero(p->fromdomain)) @@ -7396,29 +7361,29 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho /* If we're calling a registered SIP peer, use the fullcontact to dial to the peer */ if (!ast_strlen_zero(p->fullcontact)) { /* If we have full contact, trust it */ - ast_build_string(&invite, &invite_max, "%s", p->fullcontact); + ast_str_append(&invite, 0, "%s", p->fullcontact); } else { /* Otherwise, use the username while waiting for registration */ - ast_build_string(&invite, &invite_max, "sip:"); + ast_str_append(&invite, 0, "sip:"); if (!ast_strlen_zero(p->username)) { n = p->username; if (pedanticsipchecking) { - ast_uri_encode(n, tmp, sizeof(tmp), 0); - n = tmp; + ast_uri_encode(n, tmp_n, sizeof(tmp_n), 0); + n = tmp_n; } - ast_build_string(&invite, &invite_max, "%s@", n); + ast_str_append(&invite, 0, "%s@", n); } - ast_build_string(&invite, &invite_max, "%s", p->tohost); + ast_str_append(&invite, 0, "%s", p->tohost); if (ntohs(p->sa.sin_port) != STANDARD_SIP_PORT) - ast_build_string(&invite, &invite_max, ":%d", ntohs(p->sa.sin_port)); - ast_build_string(&invite, &invite_max, "%s", urioptions); + ast_str_append(&invite, 0, ":%d", ntohs(p->sa.sin_port)); + ast_str_append(&invite, 0, "%s", urioptions); } /* If custom URI options have been provided, append them */ if (p->options && p->options->uri_options) - ast_build_string(&invite, &invite_max, ";%s", p->options->uri_options); + ast_str_append(&invite, 0, ";%s", p->options->uri_options); - ast_string_field_set(p, uri, invite_buf); + ast_string_field_set(p, uri, invite->str); if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->theirtag)) { /* If this is a NOTIFY, use the From: tag in the subscribe (RFC 3265) */ @@ -7430,7 +7395,8 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho snprintf(to, sizeof(to), "<%s>", p->uri); init_req(req, sipmethod, p->uri); - snprintf(tmp, sizeof(tmp), "%d %s", ++p->ocseq, sip_methods[sipmethod].text); + /* now tmp_n is available so reuse it to build the CSeq */ + snprintf(tmp_n, sizeof(tmp_n), "%d %s", ++p->ocseq, sip_methods[sipmethod].text); add_header(req, "Via", p->via); add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS); @@ -7447,7 +7413,7 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho build_contact(p); add_header(req, "Contact", p->our_contact); add_header(req, "Call-ID", p->callid); - add_header(req, "CSeq", tmp); + add_header(req, "CSeq", tmp_n); if (!ast_strlen_zero(global_useragent)) add_header(req, "User-Agent", global_useragent); if (!ast_strlen_zero(p->rpid)) @@ -7552,9 +7518,9 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init) /*! \brief Used in the SUBSCRIBE notification subsystem */ static int transmit_state_notify(struct sip_pvt *p, int state, int full, int timeout) { - char tmp[4000], from[256], to[256]; - char *t = tmp, *c, *mfrom, *mto; - size_t maxbytes = sizeof(tmp); + struct ast_str *tmp = ast_str_alloca(4000); + char from[256], to[256]; + char *c, *mfrom, *mto; struct sip_request req; char hint[AST_MAX_EXTENSION]; char *statestring = "terminated"; @@ -7668,57 +7634,56 @@ static int transmit_state_notify(struct sip_pvt *p, int state, int full, int tim switch (p->subscribed) { case XPIDF_XML: case CPIM_PIDF_XML: - ast_build_string(&t, &maxbytes, "\n"); - ast_build_string(&t, &maxbytes, "\n"); - ast_build_string(&t, &maxbytes, "\n"); - ast_build_string(&t, &maxbytes, "\n", mfrom); - ast_build_string(&t, &maxbytes, "\n", p->exten); - ast_build_string(&t, &maxbytes, "
\n", mto); - ast_build_string(&t, &maxbytes, "\n", (local_state == NOTIFY_OPEN) ? "open" : (local_state == NOTIFY_INUSE) ? "inuse" : "closed"); - ast_build_string(&t, &maxbytes, "\n", (local_state == NOTIFY_OPEN) ? "online" : (local_state == NOTIFY_INUSE) ? "onthephone" : "offline"); - ast_build_string(&t, &maxbytes, "
\n\n\n"); + ast_str_append(&tmp, 0, + "\n" + "\n" + "\n"); + ast_str_append(&tmp, 0, "\n", mfrom); + ast_str_append(&tmp, 0, "\n", p->exten); + ast_str_append(&tmp, 0, "
\n", mto); + ast_str_append(&tmp, 0, "\n", (local_state == NOTIFY_OPEN) ? "open" : (local_state == NOTIFY_INUSE) ? "inuse" : "closed"); + ast_str_append(&tmp, 0, "\n", (local_state == NOTIFY_OPEN) ? "online" : (local_state == NOTIFY_INUSE) ? "onthephone" : "offline"); + ast_str_append(&tmp, 0, "
\n\n\n"); break; case PIDF_XML: /* Eyebeam supports this format */ - ast_build_string(&t, &maxbytes, "\n"); - ast_build_string(&t, &maxbytes, "\n", mfrom); - ast_build_string(&t, &maxbytes, "\n"); + ast_str_append(&tmp, 0, + "\n" + "\n", mfrom); + ast_str_append(&tmp, 0, "\n"); if (pidfstate[0] != '-') - ast_build_string(&t, &maxbytes, "\n", pidfstate); - ast_build_string(&t, &maxbytes, "\n"); - ast_build_string(&t, &maxbytes, "%s\n", pidfnote); /* Note */ - ast_build_string(&t, &maxbytes, "\n", p->exten); /* Tuple start */ - ast_build_string(&t, &maxbytes, "%s\n", mto); + ast_str_append(&tmp, 0, "\n", pidfstate); + ast_str_append(&tmp, 0, "\n"); + ast_str_append(&tmp, 0, "%s\n", pidfnote); /* Note */ + ast_str_append(&tmp, 0, "\n", p->exten); /* Tuple start */ + ast_str_append(&tmp, 0, "%s\n", mto); if (pidfstate[0] == 'b') /* Busy? Still open ... */ - ast_build_string(&t, &maxbytes, "open\n"); + ast_str_append(&tmp, 0, "open\n"); else - ast_build_string(&t, &maxbytes, "%s\n", (local_state != NOTIFY_CLOSED) ? "open" : "closed"); - ast_build_string(&t, &maxbytes, "\n\n"); + ast_str_append(&tmp, 0, "%s\n", (local_state != NOTIFY_CLOSED) ? "open" : "closed"); + ast_str_append(&tmp, 0, "\n\n"); break; case DIALOG_INFO_XML: /* SNOM subscribes in this format */ - ast_build_string(&t, &maxbytes, "\n"); - ast_build_string(&t, &maxbytes, "\n", p->dialogver++, full ? "full":"partial", mto); + ast_str_append(&tmp, 0, "\n"); + ast_str_append(&tmp, 0, "\n", p->dialogver++, full ? "full":"partial", mto); if ((state & AST_EXTENSION_RINGING) && global_notifyringing) - ast_build_string(&t, &maxbytes, "\n", p->exten); + ast_str_append(&tmp, 0, "\n", p->exten); else - ast_build_string(&t, &maxbytes, "\n", p->exten); - ast_build_string(&t, &maxbytes, "%s\n", statestring); + ast_str_append(&tmp, 0, "\n", p->exten); + ast_str_append(&tmp, 0, "%s\n", statestring); if (state == AST_EXTENSION_ONHOLD) { - ast_build_string(&t, &maxbytes, "\n\n" + ast_str_append(&tmp, 0, "\n\n" "\n" "\n\n", mto); } - ast_build_string(&t, &maxbytes, "\n\n"); + ast_str_append(&tmp, 0, "\n\n"); break; case NONE: default: break; } - if (t > tmp + sizeof(tmp)) - ast_log(LOG_WARNING, "Buffer overflow detected!! (Please file a bug report)\n"); - - add_header_contentLength(&req, strlen(tmp)); - add_line(&req, tmp); + add_header_contentLength(&req, tmp->used); + add_line(&req, tmp->str); return send_request(p, &req, XMIT_RELIABLE, p->ocseq); } @@ -7732,21 +7697,20 @@ static int transmit_state_notify(struct sip_pvt *p, int state, int full, int tim static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, char *vmexten) { struct sip_request req; - char tmp[500]; - char *t = tmp; - size_t maxbytes = sizeof(tmp); + struct ast_str *out = ast_str_alloca(500); initreqprep(&req, p, SIP_NOTIFY); add_header(&req, "Event", "message-summary"); add_header(&req, "Content-Type", default_notifymime); - ast_build_string(&t, &maxbytes, "Messages-Waiting: %s\r\n", newmsgs ? "yes" : "no"); - ast_build_string(&t, &maxbytes, "Message-Account: sip:%s@%s\r\n", + ast_str_append(&out, 0, "Messages-Waiting: %s\r\n", newmsgs ? "yes" : "no"); + ast_str_append(&out, 0, "Message-Account: sip:%s@%s\r\n", S_OR(vmexten, default_vmexten), S_OR(p->fromdomain, ast_inet_ntoa(p->ourip.sin_addr))); /* Cisco has a bug in the SIP stack where it can't accept the (0/0) notification. This can temporarily be disabled in sip.conf with the "buggymwi" option */ - ast_build_string(&t, &maxbytes, "Voice-Message: %d/%d%s\r\n", newmsgs, oldmsgs, (ast_test_flag(&p->flags[1], SIP_PAGE2_BUGGY_MWI) ? "" : " (0/0)")); + ast_str_append(&out, 0, "Voice-Message: %d/%d%s\r\n", + newmsgs, oldmsgs, (ast_test_flag(&p->flags[1], SIP_PAGE2_BUGGY_MWI) ? "" : " (0/0)")); if (p->subscribed) { if (p->expiry) @@ -7755,11 +7719,8 @@ static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, add_header(&req, "Subscription-State", "terminated;reason=timeout"); } - if (t > tmp + sizeof(tmp)) - ast_log(LOG_WARNING, "Buffer overflow detected!! (Please file a bug report)\n"); - - add_header_contentLength(&req, strlen(tmp)); - add_line(&req, tmp); + add_header_contentLength(&req, out->used); + add_line(&req, out->str); if (!p->initreq.headers) initialize_initreq(p, &req);