[OpenSIPS-Users] OpenSIPS FreeSWITCH TCP support
Matthias Kneer
matze at roterruler.de
Mon Jul 22 12:26:18 EDT 2019
Hello,
I'm using OpenSIPS as a loadbalancer / failover proxy and mid registrar
in AOR mode, also for doing the NAT keepalives, in front of 2 FreeSWITCHes.
While using UDP everything works fine. Since I now want to implement
TLS, I first started to test TCP to have the baseline working to go on
with TLS afterwards. Unfortunately, while using TCP, none of TCP
registered phones receives inbound calls - while outbound calls from
these phones to other UDP registered phones work without an issue. The
registration seems to work fine and also the information in the
userlocation of OpenSIPS properly recognizing the registration over TCP:
# opensipsctl ul show
Domain:: location hash_size=512
AOR:: 212 at sip.example.org
Contact::
sip:212@[EXTERNAL_PHONE_IP]:40009;transport=TCP;rinstance=ebc7149013d15489
Q=
ContactID:: 1156589880790491998
Expires:: 18
Callid:: Y6nJ2r4Iwm6WNL_KHqcLOQ..
Cseq:: 2
User-agent:: Zoiper rv2.9.2
Received::
sip:[EXTERNAL_PHONE_IP]:40009;transport=tcp
State:: CS_SYNC
Flags:: 0
Cflags::
Socket:: tcp:[OPENSIPS_IP]:5060
Methods:: 5951
It looks like FreeSWITCH is not recognizing that the phones are
registered through TCP even though the packets seem to contain the
required information. Here's a dump of a the registration through TCP
taken from the FreeSWITCH: https://pastebin.com/mbMffLK7
And here's the output of the active registrations on the FreeSWITCH for
the above mentioned phone, which also contains the transport information
in the contact header:
Call-ID: Y6nJ2r4Iwm6WNL_KHqcLOQ..
User: 212 at sip.example.org
Contact: "212 ZoIPer"
<sip:212@[OPENSIPS_IP]:5060;ctid=1156589880790491998;fs_path=sip%3A[OPENSIPS_IP]%3Btransport%3Dtcp%3Blr>
Agent: Zoiper rv2.9.2
Status: Registered(UDP)(unknown) EXP(2019-07-22 18:02:32)
EXPSECS(131)
Ping-Status: Reachable
Ping-Time: 0.00
Host: freeswitch
IP: [OPENSIPS_IP]
Port: 51540
Auth-User: 212
Auth-Realm: sip.example.org
MWI-Account: 212 at sip.example.org
Could someone guide me, at which point I do have to handle TCP different
from UDP? I'm pretty new to this whole VoIP / SIP topic.
Thanks in advance,
Matthias
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