diff options
author | Matteo Brancaleoni <mbrancaleoni@espia.it> | 2003-03-13 06:00:20 +0000 |
---|---|---|
committer | Matteo Brancaleoni <mbrancaleoni@espia.it> | 2003-03-13 06:00:20 +0000 |
commit | 4ebaef0e1cf0718219b5a31ec182582e4f31b47e (patch) | |
tree | f000e1e3d7222acbc97fd66eed3436cc5dde90e3 /channels | |
parent | 66a57e51e328a0c0b855184c15096758401c0dac (diff) |
Thu Mar 13 07:00:01 CET 2003
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
-rwxr-xr-x | channels/chan_iax2.c | 53 | ||||
-rwxr-xr-x | channels/chan_mgcp.c | 26 | ||||
-rwxr-xr-x | channels/chan_sip.c | 30 | ||||
-rwxr-xr-x | channels/iax2.h | 8 |
4 files changed, 55 insertions, 62 deletions
diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c index 7ddcf01d2..23f936efb 100755 --- a/channels/chan_iax2.c +++ b/channels/chan_iax2.c @@ -277,13 +277,13 @@ struct chan_iax2_pvt { /* Expirey (optional) */ int expirey; /* Next outgoing sequence number */ - unsigned short oseqno; + unsigned char oseqno; /* Next sequence number they have not yet acknowledged */ - unsigned short rseqno; + unsigned char rseqno; /* Next incoming sequence number */ - unsigned short iseqno; + unsigned char iseqno; /* Last incoming sequence number we have acknowledged */ - unsigned short aseqno; + unsigned char aseqno; /* Peer name */ char peer[80]; /* Default Context */ @@ -606,9 +606,9 @@ void showframe(struct ast_iax2_frame *f, struct ast_iax2_full_hdr *fhi, int rx, subclass = subclass2; } ast_verbose( -"%s-Frame Retry[%s] -- OSeqno: %4.4d ISeqno: %4.4d Type: %s Subclass: %s\n", +"%s-Frame Retry[%s] -- OSeqno: %3.3d ISeqno: %3.3d Type: %s Subclass: %s\n", (rx ? "Rx" : "Tx"), - retries, ntohs(fh->oseqno), ntohs(fh->iseqno), class, subclass); + retries, fh->oseqno, fh->iseqno, class, subclass); fprintf(stderr, " Timestamp: %05dms SCall: %5.5d DCall: %5.5d [%s:%d]\n", ntohl(fh->ts), @@ -1163,7 +1163,7 @@ static int update_packet(struct ast_iax2_frame *f) fh->dcallno = ntohs(AST_FLAG_RETRANS | f->dcallno); /* Update iseqno */ f->iseqno = iaxs[f->callno]->iseqno; - fh->iseqno = ntohs(f->iseqno); + fh->iseqno = f->iseqno; return 0; } @@ -1398,7 +1398,7 @@ static int forward_delivery(struct ast_iax2_frame *fr) } #endif -static int schedule_delivery(struct ast_iax2_frame *fr, int reallydeliver) +static int schedule_delivery(struct ast_iax2_frame *fr, int reallydeliver, int updatehistory) { int ms,x; int drops[MEMORY_SIZE]; @@ -1423,10 +1423,12 @@ static int schedule_delivery(struct ast_iax2_frame *fr, int reallydeliver) /* Rotate our history queue of "lateness". Don't worry about those initial zeros because the first entry will always be zero */ - for (x=0;x<MEMORY_SIZE - 1;x++) - iaxs[fr->callno]->history[x] = iaxs[fr->callno]->history[x+1]; - /* Add a history entry for this one */ - iaxs[fr->callno]->history[x] = ms; + if (updatehistory) { + for (x=0;x<MEMORY_SIZE - 1;x++) + iaxs[fr->callno]->history[x] = iaxs[fr->callno]->history[x+1]; + /* Add a history entry for this one */ + iaxs[fr->callno]->history[x] = ms; + } /* Initialize the minimum to reasonable values. It's too much work to do the same for the maximum, repeatedly */ @@ -2177,9 +2179,9 @@ static int iax2_send(struct chan_iax2_pvt *pvt, struct ast_frame *f, unsigned in int res; unsigned int lastsent; /* Allocate an ast_iax2_frame */ - if (now) + if (now) { fr = &fr2; - else + } else fr = ast_iax2_frame_new(DIRECTION_OUTGRESS, f->datalen); if (!fr) { ast_log(LOG_WARNING, "Out of memory\n"); @@ -2220,8 +2222,8 @@ static int iax2_send(struct chan_iax2_pvt *pvt, struct ast_frame *f, unsigned in fh = (struct ast_iax2_full_hdr *)(fr->af.data - sizeof(struct ast_iax2_full_hdr)); fh->scallno = htons(fr->callno | AST_FLAG_FULL); fh->ts = htonl(fr->ts); - fh->oseqno = htons(fr->oseqno); - fh->iseqno = htons(fr->iseqno); + fh->oseqno = fr->oseqno; + fh->iseqno = fr->iseqno; /* Keep track of the last thing we've acknowledged */ pvt->aseqno = fr->iseqno; fh->type = fr->af.frametype & 0xFF; @@ -3491,6 +3493,7 @@ static int socket_read(int *id, int fd, short events, void *cbdata) { struct sockaddr_in sin; int res; + int updatehistory=1; int new = NEW_PREVENT; char buf[4096]; int len = sizeof(sin); @@ -3568,10 +3571,13 @@ static int socket_read(int *id, int fd, short events, void *cbdata) iaxs[fr.callno]->peercallno = (short)(ntohs(mh->callno) & ~AST_FLAG_FULL); if (ntohs(mh->callno) & AST_FLAG_FULL) { if (option_debug) - ast_log(LOG_DEBUG, "Received packet %d, (%d, %d)\n", ntohs(fh->oseqno), f.frametype, f.subclass); + ast_log(LOG_DEBUG, "Received packet %d, (%d, %d)\n", fh->oseqno, f.frametype, f.subclass); /* Check if it's out of order (and not an ACK or INVAL) */ - fr.oseqno = ntohs(fh->oseqno); - fr.iseqno = ntohs(fh->iseqno); + fr.oseqno = fh->oseqno; + fr.iseqno = fh->iseqno; + fr.ts = ntohl(fh->ts); + if (ntohs(fh->dcallno) & AST_FLAG_RETRANS) + updatehistory = 0; if ((iaxs[fr.callno]->iseqno != fr.oseqno) && (iaxs[fr.callno]->iseqno || ((f.subclass != AST_IAX2_COMMAND_TXCNT) && @@ -3684,7 +3690,6 @@ static int socket_read(int *id, int fd, short events, void *cbdata) f.data = empty; memset(&ies, 0, sizeof(ies)); } - fr.ts = ntohl(fh->ts); if (f.frametype == AST_FRAME_VOICE) { if (f.subclass != iaxs[fr.callno]->voiceformat) { iaxs[fr.callno]->voiceformat = f.subclass; @@ -3713,7 +3718,7 @@ static int socket_read(int *id, int fd, short events, void *cbdata) /* Go through the motions of delivering the packet without actually doing so, unless this is a lag request since it will be done for real */ if (f.subclass != AST_IAX2_COMMAND_LAGRQ) - schedule_delivery(&fr, 0); + schedule_delivery(&fr, 0, updatehistory); switch(f.subclass) { case AST_IAX2_COMMAND_ACK: /* Do nothing */ @@ -3963,7 +3968,7 @@ static int socket_read(int *id, int fd, short events, void *cbdata) f.offset = 0; f.samples = 0; ast_iax2_frame_wrap(&fr, &f); - schedule_delivery(iaxfrdup2(&fr), 1); + schedule_delivery(iaxfrdup2(&fr), 1, updatehistory); #ifdef BRIDGE_OPTIMIZATION } #endif @@ -4239,10 +4244,10 @@ static int socket_read(int *id, int fd, short events, void *cbdata) if (iaxs[fr.callno]->bridgecallno) { forward_delivery(&fr); } else { - schedule_delivery(iaxfrdup2(&fr), 1); + schedule_delivery(iaxfrdup2(&fr), 1, updatehistory); } #else - schedule_delivery(iaxfrdup2(&fr), 1); + schedule_delivery(iaxfrdup2(&fr), 1, updatehistory); #endif /* Always run again */ ast_pthread_mutex_unlock(&iaxsl[fr.callno]); diff --git a/channels/chan_mgcp.c b/channels/chan_mgcp.c index e2a779577..0b7627d90 100755 --- a/channels/chan_mgcp.c +++ b/channels/chan_mgcp.c @@ -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); } } diff --git a/channels/chan_sip.c b/channels/chan_sip.c index d665d64e5..729bb816e 100755 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -1422,14 +1422,14 @@ static int process_sdp(struct sip_pvt *p, struct sip_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; } @@ -1438,20 +1438,14 @@ static int process_sdp(struct sip_pvt *p, struct sip_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; p->nonCodecCapability = nonCodecCapability & peerNonCodecCapability; @@ -1861,11 +1855,11 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp * if (p->capability & cur->codec) { if (sipdebug) ast_verbose("Answering with preferred capability %d\n", cur->codec); - codec = rtp_lookup_code(p->rtp, 1, cur->codec); + codec = ast_rtp_lookup_code(p->rtp, 1, cur->codec); 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, cur->codec)); + snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, cur->codec)); strcat(a, costr); } } @@ -1877,11 +1871,11 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp * if ((p->capability & x) && !(alreadysent & x)) { if (sipdebug) 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); } } @@ -1890,11 +1884,11 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp * if (p->nonCodecCapability & x) { if (sipdebug) 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... */ @@ -2894,7 +2888,7 @@ static int sip_show_channel(int fd, int argc, char *argv[]) } ast_pthread_mutex_unlock(&iflock); if (!cur) - ast_cli(fd, "No such SIP Call ID '%s'\n", cur->callid); + ast_cli(fd, "No such SIP Call ID '%s'\n", argv[3]); return RESULT_SUCCESS; } @@ -4281,7 +4275,7 @@ static int reload_config(void) } else { hp = gethostbyname(ourhost); if (!hp) { - ast_log(LOG_WARNING, "Unable to get our IP address, SIP disabled\n"); + ast_log(LOG_WARNING, "Unable to get IP address for %s, SIP disabled\n", ourhost); return 0; } memcpy(&__ourip, hp->h_addr, sizeof(__ourip)); diff --git a/channels/iax2.h b/channels/iax2.h index c3727ac81..da8acadff 100755 --- a/channels/iax2.h +++ b/channels/iax2.h @@ -107,12 +107,12 @@ struct ast_iax2_full_hdr { unsigned short scallno; /* Source call number -- high bit must be 1 */ unsigned short dcallno; /* Destination call number -- high bit is 1 if retransmission */ unsigned int ts; /* 32-bit timestamp in milliseconds (from 1st transmission) */ - unsigned short oseqno; /* Packet number (outgoing) */ - unsigned short iseqno; /* Packet number (next incoming expected) */ + unsigned char oseqno; /* Packet number (outgoing) */ + unsigned char iseqno; /* Packet number (next incoming expected) */ char type; /* Frame type */ unsigned char csub; /* Compressed subclass */ unsigned char iedata[0]; -}; +} __attribute__ ((__packed__)); /* Mini header is used only for voice frames -- delivered unreliably */ struct ast_iax2_mini_hdr { @@ -121,6 +121,6 @@ struct ast_iax2_mini_hdr { /* Frametype implicitly VOICE_FRAME */ /* subclass implicit from last ast_iax2_full_hdr */ unsigned char iedata[0]; -}; +} __attribute__ ((__packed__)); #endif |