Asterisk
/
asterisk
mirror of https://github.com/asterisk/asterisk
1
0
Fork 0
Code Issues Releases Wiki Activity

Merged revisions 109390 via svnmerge from

Browse Source 
https://origsvn.digium.com/svn/asterisk/trunk

................
r109390 | file | 2008-03-18 12:08:09 -0300 (Tue, 18 Mar 2008) | 11 lines

Merged revisions 109386 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r109386 | file | 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) | 3 lines

Put a maximum limit on the number of payloads accepted, and also make sure a given payload does not exceed our maximum value.
(AST-2008-002)

........

................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@109392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.6.0
Joshua Colp 17 years ago
parent cc4c5f1a6d
commit 627d3f04d4
2 changed files with 40 additions and 30 deletions
Show Stats Download Patch File Download Diff File

67
channels/chan_sip.c
Unescape View File

@ -243,6 +243,8 @@ static int expiry = DEFAULT_EXPIRY;
#define DEFAULT_MAX_SE 1800 /*!< Session-Timer Default Session-Expires period (RFC 4028) */
#define DEFAULT_MIN_SE 90 /*!< Session-Timer Default Min-SE period (RFC 4028) */
#define SDP_MAX_RTPMAP_CODECS 32 /*!< Maximum number of codecs allowed in received SDP */
/*! \brief Global jitterbuffer configuration - by default, jb is disabled */
static struct ast_jb_conf default_jbconf =
{
@ -6305,7 +6307,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
int numberofmediastreams = 0;
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;
char buf[SIPBUFSIZE];
@ -6655,36 +6657,41 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
} else if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) == 2) {
/* We have a rtpmap to handle */
/* Note: should really look at the 'freq' and '#chans' params too */
/* Note: This should all be done in the context of the m= above */
if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)) { /* Video */
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++;
} else {
ast_rtp_unset_m_type(newvideortp, codec);
if (debug)
ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
}
} else if (!strncasecmp(mimeSubtype, "T140", 4)) { /* Text */
if (p->trtp) {
/* ast_verbose("Adding t140 mimeSubtype to textrtp struct\n"); */
ast_rtp_set_rtpmap_type(newtextrtp, codec, "text", mimeSubtype, 0);
}
} else { /* Must be audio?? */
if(ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0) != -1) {
if (debug)
ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
found_rtpmap_codecs[last_rtpmap_codec] = codec;
last_rtpmap_codec++;
} else {
ast_rtp_unset_m_type(newaudiortp, codec);
if (debug)
ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
if (last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
/* Note: should really look at the 'freq' and '#chans' params too */
/* Note: This should all be done in the context of the m= above */
if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)) { /* Video */
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++;
} else {
ast_rtp_unset_m_type(newvideortp, codec);
if (debug)
ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
}
} else if (!strncasecmp(mimeSubtype, "T140", 4)) { /* Text */
if (p->trtp) {
/* ast_verbose("Adding t140 mimeSubtype to textrtp struct\n"); */
ast_rtp_set_rtpmap_type(newtextrtp, codec, "text", mimeSubtype, 0);
}
} else { /* Must be audio?? */
if(ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0) != -1) {
if (debug)
ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
found_rtpmap_codecs[last_rtpmap_codec] = codec;
last_rtpmap_codec++;
} else {
ast_rtp_unset_m_type(newaudiortp, codec);
if (debug)
ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
}
}
} else {
if (debug)
ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
}
}

3
main/rtp.c
Unescape View File

@ -1976,6 +1976,9 @@ void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt)
an unknown media type */
void ast_rtp_unset_m_type(struct ast_rtp* rtp, int pt)
{
if (pt < 0 || pt > MAX_RTP_PT)
return; /* bogus payload type */
rtp_bridge_lock(rtp);
rtp->current_RTP_PT[pt].isAstFormat = 0;
rtp->current_RTP_PT[pt].code = 0;

Write Preview
Loading…
Cancel
Save