dect
/
asterisk
Archived
13
0
Fork 0
Code Releases Activity

Merged revisions 132645 via svnmerge from 

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 lines

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 :-)

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@132703 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
oej 2008-07-22 20:46:11 +00:00
parent 667b602f9a
commit d942f83552
2 changed files with 129 additions and 3 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

6
channels/chan_sip.c
View File

@ -2956,11 +2956,11 @@ static int retrans_pkt(const void *data)
/* Too many retries */
if (pkt->owner && pkt->method != SIP_OPTIONS && xmitres == 0) {
if (pkt->is_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",
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,
pkt->is_fatal ? "Critical" : "Non-critical", pkt->is_resp ? "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) {
@ -2983,7 +2983,7 @@ static int retrans_pkt(const void *data)
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_with_cause(pkt->owner->owner, AST_CAUSE_PROTOCOL_ERROR);
ast_channel_unlock(pkt->owner->owner);
} else {

126
doc/sip-retransmit.txt Normal file
View File

@ -0,0 +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