kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-11-20 08:40:16 +00:00
Code Issues Packages Projects Releases Wiki Activity

Favor requested codec whenever possible (pass #1)

Browse Source 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2004-03-19 08:00:13 +00:00
parent ce9e964c5f
commit a386f01a7c
1 changed files with 26 additions and 6 deletions
Download Patch File Download Diff File Expand all files Collapse all files

32
channels/chan_sip.c
View File

@@ -206,6 +206,7 @@ static struct sip_pvt {
int needdestroy; /* if we need to be destroyed */ int needdestroy; /* if we need to be destroyed */
int capability; /* Special capability */ int capability; /* Special capability */
int jointcapability; /* Supported capability at both ends */ int jointcapability; /* Supported capability at both ends */
int prefcodec; /* Preferred codec (outbound only) */
int noncodeccapability; int noncodeccapability;
int outgoing; /* Outgoing or incoming call? */ int outgoing; /* Outgoing or incoming call? */
int authtries; /* Times we've tried to authenticate */ int authtries; /* Times we've tried to authenticate */
@@ -976,7 +977,7 @@ static int sip_pref_append(int format)
static int sip_codec_choose(int formats) static int sip_codec_choose(int formats)
{ {
struct sip_codec_pref *cur; struct sip_codec_pref *cur;
formats &= (AST_FORMAT_MAX_AUDIO - 1); formats &= ((AST_FORMAT_MAX_AUDIO << 1) - 1);
cur = prefs; cur = prefs;
while(cur) { while(cur) {
if (formats & cur->codec) if (formats & cur->codec)
@@ -2673,16 +2674,34 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
snprintf(t, sizeof(t), "t=0 0\r\n"); snprintf(t, sizeof(t), "t=0 0\r\n");
snprintf(m, sizeof(m), "m=audio %d RTP/AVP", ntohs(dest.sin_port)); snprintf(m, sizeof(m), "m=audio %d RTP/AVP", ntohs(dest.sin_port));
snprintf(m2, sizeof(m2), "m=video %d RTP/AVP", ntohs(vdest.sin_port)); snprintf(m2, sizeof(m2), "m=video %d RTP/AVP", ntohs(vdest.sin_port));
if (p->jointcapability & p->prefcodec) {
if (sipdebug)
ast_verbose("Answering/Requesting with root capability %d\n", p->prefcodec);
codec = ast_rtp_lookup_code(p->rtp, 1, p->prefcodec);
if (codec > -1) {
snprintf(costr, sizeof(costr), " %d", codec);
if (p->prefcodec <= AST_FORMAT_MAX_AUDIO) {
strncat(m, costr, sizeof(m) - strlen(m));
snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, p->prefcodec));
strncat(a, costr, sizeof(a));
} else {
strncat(m2, costr, sizeof(m2) - strlen(m2));
snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/90000\r\n", codec, ast_rtp_lookup_mime_subtype(1, p->prefcodec));
strncat(a2, costr, sizeof(a2));
}
}
alreadysent |= p->prefcodec;
}
/* Start by sending our preferred codecs */ /* Start by sending our preferred codecs */
cur = prefs; cur = prefs;
while(cur) { while(cur) {
if (p->jointcapability & cur->codec) { if (p->jointcapability & cur->codec) {
if (sipdebug) if (sipdebug)
ast_verbose("Answering with preferred capability %d\n", cur->codec); ast_verbose("Answering/Requesting with preferred capability %d\n", cur->codec);
codec = ast_rtp_lookup_code(p->rtp, 1, cur->codec); codec = ast_rtp_lookup_code(p->rtp, 1, cur->codec);
if (codec > -1) { if (codec > -1) {
snprintf(costr, sizeof(costr), " %d", codec); snprintf(costr, sizeof(costr), " %d", codec);
if (cur->codec < AST_FORMAT_MAX_AUDIO) { if (cur->codec <= AST_FORMAT_MAX_AUDIO) {
strncat(m, costr, sizeof(m) - strlen(m)); strncat(m, costr, sizeof(m) - strlen(m));
snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, cur->codec)); snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, cur->codec));
strncat(a, costr, sizeof(a)); strncat(a, costr, sizeof(a));
@@ -2704,7 +2723,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
codec = ast_rtp_lookup_code(p->rtp, 1, x); codec = ast_rtp_lookup_code(p->rtp, 1, x);
if (codec > -1) { if (codec > -1) {
snprintf(costr, sizeof(costr), " %d", codec); snprintf(costr, sizeof(costr), " %d", codec);
if (x < AST_FORMAT_MAX_AUDIO) { if (x <= AST_FORMAT_MAX_AUDIO) {
strncat(m, costr, sizeof(m) - strlen(m)); strncat(m, costr, sizeof(m) - strlen(m));
snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_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));
strncat(a, costr, sizeof(a) - strlen(a)); strncat(a, costr, sizeof(a) - strlen(a));
@@ -5972,7 +5991,7 @@ static struct ast_channel *sip_request(char *type, int format, void *data)
char *dest = data; char *dest = data;
oldformat = format; oldformat = format;
format &= capability; format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1);
if (!format) { if (!format) {
ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format %s while capability is %s\n", ast_getformatname(oldformat), ast_getformatname(capability)); ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format %s while capability is %s\n", ast_getformatname(oldformat), ast_getformatname(capability));
return NULL; return NULL;
@@ -6021,6 +6040,7 @@ static struct ast_channel *sip_request(char *type, int format, void *data)
#if 0 #if 0
printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "<none>", host); printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "<none>", host);
#endif #endif
p->prefcodec = format;
tmpc = sip_new(p, AST_STATE_DOWN, host); tmpc = sip_new(p, AST_STATE_DOWN, host);
if (!tmpc) if (!tmpc)
sip_destroy(p); sip_destroy(p);
@@ -6805,7 +6825,7 @@ int load_module()
res = reload_config(); res = reload_config();
if (!res) { if (!res) {
/* Make sure we can register our sip channel type */ /* Make sure we can register our sip channel type */
if (ast_channel_register_ex(type, tdesc, capability, sip_request, sip_devicestate)) { if (ast_channel_register_ex(type, tdesc, ((AST_FORMAT_MAX_AUDIO << 1) - 1), sip_request, sip_devicestate)) {
ast_log(LOG_ERROR, "Unable to register channel class %s\n", type); ast_log(LOG_ERROR, "Unable to register channel class %s\n", type);
return -1; return -1;
} }