[svn-commits] oej: branch 1.4 r132645 - in /branches/1.4: channels/ doc/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jul 22 15:10:26 CDT 2008
Author: oej
Date: Tue Jul 22 15:10:26 2008
New Revision: 132645
URL: http://svn.digium.com/view/asterisk?view=rev&rev=132645
Log:
The most common question on the #asterisk iRC channel and on mailing lists
seems to be in regards to an error message when retransmit fails. This
is frequently misunderstood as a failure of Asterisk, not a failure of
the network to reach the other party.
This document tries to assist the Asterisk user in sorting out these
issues by explaining the logic and pointing at some possible
causes. Hopefully, we will get other questions now :-)
Added:
branches/1.4/doc/sip-retransmit.txt (with props)
Modified:
branches/1.4/channels/chan_sip.c
Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=132645&r1=132644&r2=132645
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Tue Jul 22 15:10:26 2008
@@ -1954,9 +1954,9 @@
/* Too many retries */
if (pkt->owner && pkt->method != SIP_OPTIONS && xmitres == 0) {
if (ast_test_flag(pkt, FLAG_FATAL) || sipdebug) /* Tell us if it's critical or if we're debugging */
- ast_log(LOG_WARNING, "Maximum retries exceeded on transmission %s for seqno %d (%s %s)\n", pkt->owner->callid, pkt->seqno, (ast_test_flag(pkt, FLAG_FATAL)) ? "Critical" : "Non-critical", (ast_test_flag(pkt, FLAG_RESPONSE)) ? "Response" : "Request");
+ ast_log(LOG_WARNING, "Maximum retries exceeded on transmission %s for seqno %d (%s %s) -- See doc/sip-retransmit.txt.\n", pkt->owner->callid, pkt->seqno, (ast_test_flag(pkt, FLAG_FATAL)) ? "Critical" : "Non-critical", (ast_test_flag(pkt, FLAG_RESPONSE)) ? "Response" : "Request");
} else if ((pkt->method == SIP_OPTIONS) && sipdebug) {
- ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) \n", pkt->owner->callid);
+ ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) -- See doc/sip-retransmit.txt.\n", pkt->owner->callid);
}
if (xmitres == XMIT_ERROR) {
ast_log(LOG_WARNING, "Transmit error :: Cancelling transmission of transaction in call id %s \n", pkt->owner->callid);
@@ -1976,7 +1976,7 @@
if (pkt->owner->owner) {
sip_alreadygone(pkt->owner);
- ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet.\n", pkt->owner->callid);
+ ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet (see doc/sip-retransmit.txt).\n", pkt->owner->callid);
ast_queue_hangup(pkt->owner->owner);
ast_channel_unlock(pkt->owner->owner);
} else {
Added: branches/1.4/doc/sip-retransmit.txt
URL: http://svn.digium.com/view/asterisk/branches/1.4/doc/sip-retransmit.txt?view=auto&rev=132645
==============================================================================
--- branches/1.4/doc/sip-retransmit.txt (added)
+++ branches/1.4/doc/sip-retransmit.txt Tue Jul 22 15:10:26 2008
@@ -1,0 +1,126 @@
+What is the problem with SIP retransmits?
+-----------------------------------------
+
+Sometimes you get messages in the console like these:
+
+- "retrans_pkt: Hanging up call XX77yy - no reply to our critical packet."
+- "retrans_pkt: Cancelling retransmit of OPTIONs"
+
+The SIP protocol is based on requests and replies. Both sides send
+requests and wait for replies. Some of these requests are important.
+In a TCP/IP network many things can happen with IP packets. Firewalls,
+NAT devices, Session Border Controllers and SIP Proxys are in the
+signalling path and they will affect the call.
+
+SIP Call setup - INVITE-200 OK - ACK
+------------------------------------
+To set up a SIP call, there's an INVITE transaction. The SIP software that
+initiates the call sends an INVITE, then wait to get a reply. When a
+reply arrives, the caller sends an ACK. This is a three-way handshake
+that is in place since a phone can ring for a very long time and
+the protocol needs to make sure that all devices are still on line
+when call setup is done and media starts to flow.
+
+- The first reply we're waiting for is often a "100 trying".
+ This message means that some type of SIP server has received our
+ request and makes sure that we will get a reply. It could be
+ the other endpoint, but it could also be a SIP proxy or SBC
+ that handles the request on our behalf.
+
+- After that, you often see a response in the 18x class, like
+ "180 ringing" or "183 Session Progress". This typically means that our
+ request has reached at least one endpoint and something
+ is alerting the other end that there's a call coming in.
+
+- Finally, the other side answers and we get a positive reply,
+ "200 OK". This is a positive answer. In that message, we get an
+ address that goes directly to the device that answers. Remember,
+ there could be multiple phones ringing. The address is specified
+ by the Contact: header.
+
+- To confirm that we can reach the phone that answered our call,
+ we now send an ACK to the Contact: address. If this ACK doesn't
+ reach the phone, the call fails. If we can't send an ACK, we
+ can't send anything else, not even a proper hangup. Call
+ signalling will simply fail for the rest of the call and there's
+ no point in keeping it alive.
+
+- If we get an error response to our INVITE, like "Busy" or
+ "Rejected", we send the ACK to the same address as we sent the
+ INVITE, to confirm that we got the response.
+
+In order to make sure that the whole call setup sequence works and that
+we have a call, a SIP client retransmits messages if there's too much
+delay between request and expected response. We retransmit a number of
+times while waiting for the first response. We retransmit the answer to an
+incoming INVITE while waiting for an ACK. If we get multiple answers,
+we send an ACK to each of them.
+
+If we don't get the ACK or don't get an answer to our INVITE,
+even after retransmissions, we will hangup the call with the first
+error message you see above.
+
+Other SIP requests
+------------------
+Other SIP requests are only based on request - reply. There's
+no ACK, no three-way handshake. In Asterisk we mark some of
+these as CRITICAL - they need to go through for the call to
+work as expected. Some are non-critical, we don't really care
+what happens with them, the call will go on happily regardless.
+
+The qualification process - OPTIONS
+-----------------------------------
+If you turn on qualify= in sip.conf for a device, Asterisk will
+send an OPTIONS request every minute to the device and check
+if it replies. Each OPTIONS request is retransmitted a number
+of times (to handle packet loss) and if we get no reply, the
+device is considered unreachable. From that moment, we will
+send a new OPTIONS request (with retransmits) every tenth
+second.
+
+Why does this happen?
+---------------------
+
+For some reason signalling doesn't work as expected between
+your Asterisk server and the other device. There could be many reasons
+why this happens.
+
+- A NAT device in the signalling path
+ A misconfigured NAT device is in the signalling path
+ and stops SIP messages.
+- A firewall that blocks messages or reroutes them wrongly
+ in an attempt to assist in a too clever way.
+- A SIP middlebox (SBC) that rewrites contact: headers
+ so that we can't reach the other side with our reply
+ or the ACK.
+- A badly configured SIP proxy that forgets to add
+ record-route headers to make sure that signalling works.
+- Packet loss. IP and UDP are unreliable transports. If
+ you loose too many packets the retransmits doesn't help
+ and communication is impossible. If this happens with
+ signalling, media would be unusable anyway.
+
+What can I do?
+--------------
+
+Turn on SIP debug, try to understand the signalling that happens
+and see if you're missing the reply to the INVITE or if the
+ACK gets lost. When you know what happens, you've taken the
+first step to track down the problem. See the list above and
+investigate your network.
+
+For NAT and Firewall problems, there are many documents
+to help you. Start with reading sip.conf.sample that is
+part of your Asterisk distribution.
+
+The SIP signalling standard, including retransmissions
+and timers for these, is well documented in the IETF
+RFC 3261.
+
+Good luck sorting out your SIP issues!
+
+/Olle E. Johansson
+
+
+-- oej (at) edvina.net, Sweden, 2008-07-22
+-- http://www.voip-forum.com
Propchange: branches/1.4/doc/sip-retransmit.txt
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