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Thu Mar 13 07:00:01 CET 2003

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Matteo Brancaleoni
2003-03-13 06:00:20 +00:00
parent 66a57e51e3
commit 4ebaef0e1c
6 changed files with 94 additions and 82 deletions
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26
channels/chan_mgcp.c
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@@ -867,14 +867,14 @@ static int process_sdp(struct mgcp_endpoint *p, struct mgcp_request *req)
printf("Peer RTP is at port %s:%d\n", inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
#endif
// Scan through the RTP payload types specified in a "m=" line:
rtp_pt_init(p->rtp);
ast_rtp_pt_clear(p->rtp);
codecs = m + len;
while(strlen(codecs)) {
if (sscanf(codecs, "%d %n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
return -1;
}
rtp_set_m_type(p->rtp, codec);
ast_rtp_set_m_type(p->rtp, codec);
codecs += len;
}
@@ -883,20 +883,14 @@ static int process_sdp(struct mgcp_endpoint *p, struct mgcp_request *req)
sdpLineNum_iterator_init(&iterator);
while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
char* mimeSubtype = strdup(a); // ensures we have enough space
int subtypeLen, i;
if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2) continue;
// Note: should really look at the 'freq' and '#chans' params too
subtypeLen = strlen(mimeSubtype);
// Convert the MIME subtype to upper case, for ease of searching:
for (i = 0; i < subtypeLen; ++i) {
mimeSubtype[i] = toupper(mimeSubtype[i]);
}
rtp_set_rtpmap_type(p->rtp, codec, "audio", mimeSubtype);
ast_rtp_set_rtpmap_type(p->rtp, codec, "audio", mimeSubtype);
free(mimeSubtype);
}
// Now gather all of the codecs that were asked for:
rtp_get_current_formats(p->rtp,
ast_rtp_get_current_formats(p->rtp,
&peercapability, &peerNonCodecCapability);
p->capability = capability & peercapability;
if (mgcpdebug) {
@@ -1065,11 +1059,11 @@ static int add_sdp(struct mgcp_request *resp, struct mgcp_endpoint *p, struct as
if (p->capability & x) {
if (mgcpdebug)
ast_verbose("Answering with capability %d\n", x);
codec = rtp_lookup_code(p->rtp, 1, x);
codec = ast_rtp_lookup_code(p->rtp, 1, x);
if (codec > -1) {
snprintf(costr, sizeof(costr), " %d", codec);
strcat(m, costr);
snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, rtp_lookup_mime_subtype(1, x));
snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, x));
strcat(a, costr);
}
}
@@ -1078,11 +1072,11 @@ static int add_sdp(struct mgcp_request *resp, struct mgcp_endpoint *p, struct as
if (p->nonCodecCapability & x) {
if (mgcpdebug)
ast_verbose("Answering with non-codec capability %d\n", x);
codec = rtp_lookup_code(p->rtp, 0, x);
codec = ast_rtp_lookup_code(p->rtp, 0, x);
if (codec > -1) {
snprintf(costr, sizeof(costr), " %d", codec);
strcat(m, costr);
snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, rtp_lookup_mime_subtype(0, x));
snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(0, x));
strcat(a, costr);
if (x == AST_RTP_DTMF) {
/* Indicate we support DTMF... Not sure about 16, but MSN supports it so dang it, we will too... */
@@ -1121,7 +1115,7 @@ static int transmit_modify_with_sdp(struct mgcp_endpoint *p, struct ast_rtp *rtp
snprintf(local, sizeof(local), "p:20");
for (x=1;x<= AST_FORMAT_MAX_AUDIO; x <<= 1) {
if (p->capability & x) {
snprintf(tmp, sizeof(tmp), ", a:%s", rtp_lookup_mime_subtype(1, x));
snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x));
strcat(local, tmp);
}
}
@@ -1146,7 +1140,7 @@ static int transmit_connect_with_sdp(struct mgcp_endpoint *p, struct ast_rtp *rt
snprintf(local, sizeof(local), "p:20");
for (x=1;x<= AST_FORMAT_MAX_AUDIO; x <<= 1) {
if (p->capability & x) {
snprintf(tmp, sizeof(tmp), ", a:%s", rtp_lookup_mime_subtype(1, x));
snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x));
strcat(local, tmp);
}
}