| Summary: | ASTERISK-17750: DTMF transmission does not meet the recommendations of the rfc2833(4733). | ||||
| Reporter: | Mike Danilenko (muhlik) | Labels: | |||
| Date Opened: | 2011-04-25 08:51:17 | Date Closed: | 2012-09-27 12:00:11 | ||
| Priority: | Minor | Regression? | Yes | ||
| Status: | Closed/Complete | Components: | Core/RTP | ||
| Versions: | 1.8.3 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | Attachments: | ( 0) issue17750-rtp-seqnum-trunk361279.diff ( 1) RTP_RCF2833_end_seqno.patch | |||
| Description: | According to rfc4733: {code} Network Working Group H. Schulzrinne Request for Comments: 4733 Columbia U. Obsoletes: 2833 T. Taylor Category: Standards Track Nortel December 2006 2.5.1.6. RTP Sequence Number The RTP sequence number MUST be incremented by one in each successive RTP packet sent. Incrementing applies to retransmitted as well as initial instances of event reports, to permit the receiver to detect lost packets for RTP Control Protocol (RTCP) receiver reports. {code} We can see in res_rtp_asterisk.c {code} 644: static int ast_rtp_dtmf_end_with_duration(struct ast_rtp_instance *instance, char digit, unsigned int duration) ... 692: for (i = 0; i < 3; i++) { 693: res = rtp_sendto(instance, (void *) rtpheader, hdrlen + 4, 0, &remote_address); 694: if (res < 0) { 695: ast_log(LOG_ERROR, "RTP Transmission error to %s: %s\n", 696: ast_sockaddr_stringify(&remote_address), strerror(errno)); 698: } 699: if (rtp_debug_test_addr(&remote_address)) { 700: ast_verbose("Sent RTP DTMF packet to %s (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n", 701: ast_sockaddr_stringify(&remote_address), 702: rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen); 703: } 704: } {code} in this function rtp->seqno++ is absent In 1.6.2.17.3 version * in file rtp.c: {code} 3295: int ast_rtp_senddigit_end_with_duration(struct ast_rtp *rtp, char digit, unsigned int duration) ... 3336: for (i = 0; i < 3; i++) { 3337: rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno)); 3338: res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them)); 3339: rtp->seqno++; 3340: if (res < 0) 3341: ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n", 3342: ast_inet_ntoa(rtp->them.sin_addr), 3343: ntohs(rtp->them.sin_port), strerror(errno)); 3344: if (rtp_debug_test_addr(&rtp->them)) 3345: ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n", 3346: ast_inet_ntoa(rtp->them.sin_addr), 3347: ntohs(rtp->them.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen); 3348: } {code} Some SIP-gateway work incorrectly in this situation. If you are using Dial() with option D(some_number), SIP-gateway taking only the first tone. ****** ADDITIONAL INFORMATION ****** This problem is similar to: ASTERISK-12308 (https://issues.asterisk.org/view.php?id=12983) and apply to 1.8.3.2, 1.8.4 and may be to all 1.8.x versions | ||||
| Comments: | By: Mike Danilenko (muhlik) 2011-04-26 08:47:58 my patch: --- res/res_rtp_asterisk.c 2010-10-06 08:35:51.000000000 +0400 +++ res/res_rtp_asterisk.c 2011-04-26 17:39:55.237263918 +0400 @@ -686,10 +686,10 @@ rtpheader[2] = htonl(rtp->ssrc); rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration)); rtpheader[3] |= htonl((1 << 23)); - rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno)); /* Send it 3 times, that's the magical number */ for (i = 0; i < 3; i++) { + rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno)); res = rtp_sendto(instance, (void *) rtpheader, hdrlen + 4, 0, &remote_address); if (res < 0) { ast_log(LOG_ERROR, "RTP Transmission error to %s: %s\n", @@ -701,6 +701,7 @@ ast_sockaddr_stringify(&remote_address), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen); } + rtp->seqno++; } /* Oh and we can't forget to turn off the stuff that says we are sending DTMF */ By: Igor Goncharovsky (igorg) 2012-04-06 03:02:21.977-0500 Confirmed, this issue exists in 1.8, 10 and trunk. Patch for current trunk attached. By: Filip Frank (frenk77) 2012-06-22 02:14:41.832-0500 I have a same problem, if i fix it DTMF works well now. I attached my patch, but is similar as patch from Mike. I must past it here because add attachment writes me some license problem. --- res_rtp_asterisk.c 2012-06-21 13:41:34.780470962 +0200 +++ res_rtp_asterisk.c 2012-06-21 20:54:15.530623741 +0200 @@ -802,6 +802,8 @@ ast_sockaddr_stringify(&remote_address), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen); } + rtp->seqno++; + rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno)); } /* Oh and we can't forget to turn off the stuff that says we are sending DTMF */ By: Jan Červenka (schod) 2012-06-22 03:38:12.980-0500 Hi, I tested Filip's patch on Asterisk 1.8.13.0 and it solved my problem with DTMF. Thx By: David Woolley (davidw) 2012-06-22 04:29:07.572-0500 Patches must be attached and you must first have agreed to the licence terms. You are granting Digium permission to use the patch in their commercial products, which you do by following the Sign a Licence Agreement link, at the top of the web page. Unless you submit the patch as an attachment, with a valid licence grant, it will never be used in the official version. By: Filip Frank (frenk77) 2012-06-25 13:36:38.030-0500 Ok I have a signed license now. :) Patch is now attached. Please merge it to official versions, it helps with many strange DTMF problems. By: Filip Frank (frenk77) 2012-07-09 02:56:34.276-0500 It is two week after i send my patch and no response. What is a problem ? By: Igor Goncharovsky (igorg) 2012-07-09 03:31:42.921-0500 The main reason for it that issue is not assigned to anybody. Try to ask on #asterisk-dev IRC channel for anyone to look ad this. By: Matt Jordan (mjordan) 2012-09-27 12:00:01.085-0500 Fixed in r372199 | ||||