You could use iptables or a sip proxy to rewrite it. On Wed, Aug 22, 2012 at 8:37 AM, Vladimir Ananiev <ananiev@svyaz.com> wrote:
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@sip.test.com>;tag=25ffcc01-d007-1910-91fb-1cc1de60bc3c Call-ID: 25ffcc01-d007-1910-91fa-1cc1de60bc3c To: <sip:621@sip.test.com> Contact: <sip:621@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@svyaz.com
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