[asterisk-users] INVITE retransmission by 1.8... (Steve Murphy)
sst at sil.at
Mon Mar 19 03:55:41 CDT 2012
Am 18.03.12 19:53, schrieb Freddi Hansen:
>> I have a site that moved to the latest 1.8 revision, and began to
>> have problems with phones in "far away places" (South America,
>> and the MidEast).
>> What I see is that when a Dial() is issued, the sip channel driver
>> sends out an INVITE to the phone. Very soon thereafter, Asterisk
>> retransmits the INVITE. The phone sends back a 100 Trying, and
>> then, usually, a 400 response. I may be misinterpreting what I see,
>> but I *think* that the response from the phone is delayed just enough
>> to invoke the retransmission. The phone responds to the second invite
>> with a 400 code, which Asterisk interprets as an error, and the call
>> goes into hangup.
>> Has anyone else seen this? It doesn't happen all the time, and only
>> with certain
>> phones. It comes and goes, but when it comes, phones become
>> unreachable. It
>> seems to be in this state the majority of the time for certain phones.
>> While most
>> phones seem far away, some are in Florida.
>> We replaced the 1.8 version of Asterisk with a 1.6.2 version, and the
>> issue has
>> gone away. I know, I know, I could give more detail, fill out a bug
>> report, but
>> this is the early stages. I may be misinterpreting what I'm seeing.
>> Anyone else seen this sort of thing? Any words of wisdom?
There is a option in sip.conf you should use for this and its called T1
the normal retransmit timeout is either T1*2 counting up for each
retransmission (T1*4,*8,...T1*64) or it depends on the maxms values
which a qualify gives you.
So if you enable qualify and you have values like 300 ms to the phone
the retransmission should honor this, or as i said just change the t1
value from 500 ms (default) to 1000 or 2000 ms for example.
then asterisk should not try to retransmit the next invite so early.
> one of our gateways is used for SIP over satelite links and we se the
> same thing on default installation.
> The fix is to change chan_sip.c
> #define DEFAULT_RETRANS to a higher value, we use 3000.
> The retransmit timer at the far end (pap2t) is increased to 3 times its
> standard values.
> It probably breaks some sip specs but its needed to keep it working when
> roundtrip gets to big.
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