From cbdb2dbb0e25f7ab23379b02467b055e263d345b Mon Sep 17 00:00:00 2001 From: Joshua Colp Date: Wed, 18 Jul 2012 11:38:05 +0000 Subject: Fix a crash occurring as a result of excess stack usage. This fix involves moving the allocation of some temporary codec structures to the heap and also reduces the number of maximum payloads to something more sane for both regular and low memory builds. (closes issue ASTERISK-20140) Reported by: jonnt git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370171 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- channels/chan_sip.c | 50 ++++++++++++++++++++++++++++++-------------------- 1 file changed, 30 insertions(+), 20 deletions(-) (limited to 'channels') diff --git a/channels/chan_sip.c b/channels/chan_sip.c index cbf88113c..eadf9b961 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -9367,7 +9367,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action int peernoncodeccapability = 0, vpeernoncodeccapability = 0, tpeernoncodeccapability = 0; - struct ast_rtp_codecs newaudiortp, newvideortp, newtextrtp; + struct ast_rtp_codecs *newaudiortp = NULL, *newvideortp = NULL, *newtextrtp = NULL; struct ast_format_cap *newjointcapability = ast_format_cap_alloc_nolock(); /* Negotiated capability */ struct ast_format_cap *newpeercapability = ast_format_cap_alloc_nolock(); int newnoncodeccapability; @@ -9404,10 +9404,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action goto process_sdp_cleanup; } - /* Make sure that the codec structures are all cleared out */ - ast_rtp_codecs_payloads_clear(&newaudiortp, NULL); - ast_rtp_codecs_payloads_clear(&newvideortp, NULL); - ast_rtp_codecs_payloads_clear(&newtextrtp, NULL); + if (!(newaudiortp = ast_calloc(1, sizeof(*newaudiortp))) || !(newvideortp = ast_calloc(1, sizeof(*newvideortp))) || + !(newtextrtp = ast_calloc(1, sizeof(*newtextrtp)))) { + res = -1; + goto process_sdp_cleanup; + } /* Update our last rtprx when we receive an SDP, too */ p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */ @@ -9448,11 +9449,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action if (process_sdp_a_sendonly(value, &sendonly)) { processed = TRUE; } - else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec)) + else if (process_sdp_a_audio(value, p, newaudiortp, &last_rtpmap_codec)) processed = TRUE; - else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec)) + else if (process_sdp_a_video(value, p, newvideortp, &last_rtpmap_codec)) processed = TRUE; - else if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) + else if (process_sdp_a_text(value, p, newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) processed = TRUE; else if (process_sdp_a_image(value, p)) processed = TRUE; @@ -9566,7 +9567,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action ast_verbose("Found RTP audio format %d\n", codec); } - ast_rtp_codecs_payloads_set_m_type(&newaudiortp, NULL, codec); + ast_rtp_codecs_payloads_set_m_type(newaudiortp, NULL, codec); } } else { ast_log(LOG_WARNING, "Rejecting audio media offer due to invalid or unsupported syntax: %s\n", m); @@ -9638,7 +9639,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action if (debug) { ast_verbose("Found RTP video format %d\n", codec); } - ast_rtp_codecs_payloads_set_m_type(&newvideortp, NULL, codec); + ast_rtp_codecs_payloads_set_m_type(newvideortp, NULL, codec); } } else { ast_log(LOG_WARNING, "Rejecting video media offer due to invalid or unsupported syntax: %s\n", m); @@ -9702,7 +9703,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action if (debug) { ast_verbose("Found RTP text format %d\n", codec); } - ast_rtp_codecs_payloads_set_m_type(&newtextrtp, NULL, codec); + ast_rtp_codecs_payloads_set_m_type(newtextrtp, NULL, codec); } } else { ast_log(LOG_WARNING, "Rejecting text stream offer due to invalid or unsupported syntax: %s\n", m); @@ -9820,7 +9821,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action } else if (!processed_crypto && process_crypto(p, p->rtp, &p->srtp, value)) { processed_crypto = TRUE; processed = TRUE; - } else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec)) { + } else if (process_sdp_a_audio(value, p, newaudiortp, &last_rtpmap_codec)) { processed = TRUE; } } @@ -9831,7 +9832,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action } else if (!processed_crypto && process_crypto(p, p->vrtp, &p->vsrtp, value)) { processed_crypto = TRUE; processed = TRUE; - } else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec)) { + } else if (process_sdp_a_video(value, p, newvideortp, &last_rtpmap_codec)) { processed = TRUE; } } @@ -9839,7 +9840,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action else if (text) { if (process_sdp_a_ice(value, p, p->trtp)) { processed = TRUE; - } if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) { + } if (process_sdp_a_text(value, p, newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) { processed = TRUE; } else if (!processed_crypto && process_crypto(p, p->trtp, &p->tsrtp, value)) { processed_crypto = TRUE; @@ -9912,9 +9913,9 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action } /* Now gather all of the codecs that we are asked for: */ - ast_rtp_codecs_payload_formats(&newaudiortp, peercapability, &peernoncodeccapability); - ast_rtp_codecs_payload_formats(&newvideortp, vpeercapability, &vpeernoncodeccapability); - ast_rtp_codecs_payload_formats(&newtextrtp, tpeercapability, &tpeernoncodeccapability); + ast_rtp_codecs_payload_formats(newaudiortp, peercapability, &peernoncodeccapability); + ast_rtp_codecs_payload_formats(newvideortp, vpeercapability, &vpeernoncodeccapability); + ast_rtp_codecs_payload_formats(newtextrtp, tpeercapability, &tpeernoncodeccapability); ast_format_cap_append(newpeercapability, peercapability); ast_format_cap_append(newpeercapability, vpeercapability); @@ -9977,7 +9978,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action ast_sockaddr_stringify(sa)); } - ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp); + ast_rtp_codecs_payloads_copy(newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp); /* Ensure RTCP is enabled since it may be inactive if we're coming back from a T.38 session */ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1); @@ -10024,7 +10025,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action ast_verbose("Peer video RTP is at port %s\n", ast_sockaddr_stringify(vsa)); } - ast_rtp_codecs_payloads_copy(&newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp); + ast_rtp_codecs_payloads_copy(newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp); } else { ast_rtp_instance_stop(p->vrtp); if (debug) @@ -10048,7 +10049,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action } else { p->red = 0; } - ast_rtp_codecs_payloads_copy(&newtextrtp, ast_rtp_instance_get_codecs(p->trtp), p->trtp); + ast_rtp_codecs_payloads_copy(newtextrtp, ast_rtp_instance_get_codecs(p->trtp), p->trtp); } else { ast_rtp_instance_stop(p->trtp); if (debug) @@ -10166,6 +10167,15 @@ process_sdp_cleanup: if (res) { offered_media_list_destroy(p); } + if (newtextrtp) { + ast_free(newtextrtp); + } + if (newvideortp) { + ast_free(newvideortp); + } + if (newaudiortp) { + ast_free(newaudiortp); + } ast_format_cap_destroy(peercapability); ast_format_cap_destroy(vpeercapability); ast_format_cap_destroy(tpeercapability); -- cgit v1.2.3