Wrong transport=tcp in SIP Contact field.
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
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Thanks for the idea, I will try to use iptables for this. 22.08.2012, в 18:43, Anthony Minessale <anthony.minessale@gmail.com> написал(а):
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (2)
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Anthony Minessale
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Vladimir Ananiev