[Freeswitch-users] Wrong transport=tcp in SIP Contact field.
Vladimir Ananiev
ananiev at svyaz.com
Wed Aug 22 17:37:18 MSD 2012
Hello all.
I have a problem with some specific softphone connecting to the Freswitch system.
This softphone connects with UDP protocol, but sends additional ;transport=tcp line in the end of Contact field, that makes freeswitch thinking that the transport is really TCP, not the UDP. And after that incoming calls are not working. Outgoing calls from that softphone are working just fine.
Is there a way to force freeswitch to use UDP in this case ? I already have <param name="bind-params" value="transport=udp" /> in sip profile settings...
See the packet dump from wireshark:
User Datagram Protocol, Src Port: na-localise (5062), Dst Port: sip (5060)
Source port: na-localise (5062)
Destination port: sip (5060)
Length: 793
Checksum: 0x34ff [validation disabled]
Session Initiation Protocol
Request-Line: REGISTER sip:sip.test.com SIP/2.0
Method: REGISTER
Request-URI: sip:sip.test.com
[Resent Packet: False]
Message Header
CSeq: 535 REGISTER
Via: SIP/2.0/UDP 192.168.0.12:5062;branch=z9hG4bK2edb4d2b-d007-1910-9210-1cc1de60bc3c;rport
User-Agent: TMSoftPhone
[truncated] Authorization: Digest username="621", realm="sip.test.com", nonce="c4f876ae-eb58-11e1-8919-6df0d0a4f1d4", uri="sip:sip.test.com", algorithm=MD5, response="52625726279aaf004b61871164b61088", cnonce="0560cd01-d007-1910-91f
From: <sip:621 at sip.test.com>;tag=25ffcc01-d007-1910-91fb-1cc1de60bc3c
Call-ID: 25ffcc01-d007-1910-91fa-1cc1de60bc3c
To: <sip:621 at sip.test.com>
Contact: <sip:621 at 192.168.0.12:5062;transport=tcp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Expires: 120
Content-Length: 0
Max-Forwards: 70
---
Ананьев Владимир Викторович
Технический Директор
ЗАО "СВЯЗЬ"
Тел. +7 495 7238000, доб. 101
Факс.+7 495 7308111
E-mail: ananiev at svyaz.com
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