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Put a maximum limit on the number of payloads accepted, and also make sure a given payload does not exceed our maximum value.

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(AST-2008-002)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@109386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2008-03-18 14:58:39 +00:00
parent 7f77c58ed5
commit 5fda7910c6
2 changed files with 24 additions and 13 deletions
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34
channels/chan_sip.c
View File

@@ -216,6 +216,8 @@ static int expiry = DEFAULT_EXPIRY;
#define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */ #define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */
#define SIP_MAX_PACKET 4096 /*!< Also from RFC 3261 (2543), should sub headers tho */ #define SIP_MAX_PACKET 4096 /*!< Also from RFC 3261 (2543), should sub headers tho */
#define SDP_MAX_RTPMAP_CODECS 32 /*!< Maximum number of codecs allowed in received SDP */
#define INITIAL_CSEQ 101 /*!< our initial sip sequence number */ #define INITIAL_CSEQ 101 /*!< our initial sip sequence number */
/*! \brief Global jitterbuffer configuration - by default, jb is disabled */ /*! \brief Global jitterbuffer configuration - by default, jb is disabled */
@@ -5032,7 +5034,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
int numberofmediastreams = 0; int numberofmediastreams = 0;
int debug = sip_debug_test_pvt(p); int debug = sip_debug_test_pvt(p);
int found_rtpmap_codecs[32]; int found_rtpmap_codecs[SDP_MAX_RTPMAP_CODECS];
int last_rtpmap_codec=0; int last_rtpmap_codec=0;
if (!p->rtp) { if (!p->rtp) {
@@ -5305,24 +5307,30 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
/* We should propably check if this is an audio or video codec /* We should propably check if this is an audio or video codec
so we know where to look */ so we know where to look */
/* Note: should really look at the 'freq' and '#chans' params too */ if (last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
if(ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype, /* Note: should really look at the 'freq' and '#chans' params too */
ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0) != -1) { if(ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
if (debug) ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0) != -1) {
ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
found_rtpmap_codecs[last_rtpmap_codec] = codec;
last_rtpmap_codec++;
found = TRUE;
} else if (p->vrtp) {
if(ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0) != -1) {
if (debug) if (debug)
ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec); ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
found_rtpmap_codecs[last_rtpmap_codec] = codec; found_rtpmap_codecs[last_rtpmap_codec] = codec;
last_rtpmap_codec++; last_rtpmap_codec++;
found = TRUE; found = TRUE;
} else if (p->vrtp) {
if(ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0) != -1) {
if (debug)
ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec);
found_rtpmap_codecs[last_rtpmap_codec] = codec;
last_rtpmap_codec++;
found = TRUE;
}
} }
} else {
if (debug)
ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
} }
if (!found) { if (!found) {
/* Remove this codec since it's an unknown media type for us */ /* Remove this codec since it's an unknown media type for us */
/* XXX This is buggy since the media line for audio and video can have the /* XXX This is buggy since the media line for audio and video can have the

3
main/rtp.c
View File

@@ -1652,6 +1652,9 @@ void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt)
an unknown media type */ an unknown media type */
void ast_rtp_unset_m_type(struct ast_rtp* rtp, int pt) void ast_rtp_unset_m_type(struct ast_rtp* rtp, int pt)
{ {
if (pt < 0 || pt > MAX_RTP_PT)
return; /* bogus payload type */
ast_mutex_lock(&rtp->bridge_lock); ast_mutex_lock(&rtp->bridge_lock);
rtp->current_RTP_PT[pt].isAstFormat = 0; rtp->current_RTP_PT[pt].isAstFormat = 0;
rtp->current_RTP_PT[pt].code = 0; rtp->current_RTP_PT[pt].code = 0;