I forgot to add that it fails after timeout, it is maybe important. Besides, I've tested with a sip server that doesn't support tcp but refuse directly connection and fallback works properly. But in case of timeout it doesn't retry with UDP as transport. Below the log I got, as you see, there is only the tcp timeout and then no try is done on UDP. First is the UDP first invite, then we have a 407 that requires auth, and then as the next message is longer than 1300, so it use the TCP transport (that is what I understand from logs). Finally TCP timeout and the call end. However, it is maybe a limitation due to the fact it is a timeout and not a clean refuse? But it would probably make sense to try anyway on UDP : as I said, if I attach the account to the UDP transport it works fine. 01-08 04:38:40.453: DEBUG/libpjsip(598): 04:38:40.463 pjsua_core.c TX 1110 bytes Request msg INVITE/cseq=20133 (tdta0x3dc238) to UDP 212.27.52.5:5060: 01-08 04:38:40.453: DEBUG/libpjsip(598): INVITE sip:04xxxxxx at freephonie.net SIP/2.0 01-08 04:38:40.453: DEBUG/libpjsip(598): Via: SIP/2.0/UDP 83.206.185.81:1113;rport;branch=z9hG4bKPjGMc7IBRp1fZEpPMmLu1fko8JALVv69na 01-08 04:38:40.453: DEBUG/libpjsip(598): Max-Forwards: 70 01-08 04:38:40.453: DEBUG/libpjsip(598): From: <sip:09xxxxxxxx at freephonie.net>;tag=4z92T2TUU-15F1BmZ.9t-cOOV-x9FlkJ 01-08 04:38:40.453: DEBUG/libpjsip(598): To: sip:04xxxxxxxx at freephonie.net 01-08 04:38:40.453: DEBUG/libpjsip(598): Contact: <sip:0953949016 at 83.206.185.81:1113;transport=UDP> 01-08 04:38:40.453: DEBUG/libpjsip(598): Call-ID: Oe-.NHAmoQNc44Njh6sMgx7LfUwypFAX 01-08 04:38:40.453: DEBUG/libpjsip(598): CSeq: 20133 INVITE 01-08 04:38:40.453: DEBUG/libpjsip(598): Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS 01-08 04:38:40.453: DEBUG/libpjsip(598): Supported: replaces, 100rel, timer, norefersub 01-08 04:38:40.453: DEBUG/libpjsip(598): Session-Expires: 1800 01-08 04:38:40.453: DEBUG/libpjsip(598): Min-SE: 90 01-08 04:38:40.453: DEBUG/libpjsip(598): User-Agent: CSipSimple 01-08 04:38:40.453: DEBUG/libpjsip(598): Content-Type: application/sdp 01-08 04:38:40.453: DEBUG/libpjsip(598): Content-Length: 465 01-08 04:38:40.453: DEBUG/libpjsip(598): 01-08 04:38:40.453: DEBUG/libpjsip(598): v=0 01-08 04:38:40.453: DEBUG/libpjsip(598): o=- 2525139520 2525139520 IN IP4 172.20.254.194 01-08 04:38:40.453: DEBUG/libpjsip(598): s=pjmedia 01-08 04:38:40.453: DEBUG/libpjsip(598): c=IN IP4 172.20.254.194 01-08 04:38:40.453: DEBUG/libpjsip(598): t=0 0 01-08 04:38:40.453: DEBUG/libpjsip(598): a=X-nat:0 01-08 04:38:40.453: DEBUG/libpjsip(598): m=audio 4000 RTP/AVP 0 8 102 109 9 3 104 103 101 01-08 04:38:40.453: DEBUG/libpjsip(598): a=rtcp:4001 IN IP4 172.20.254.194 01-08 04:38:40.453: DEBUG/libpjsip(598): a=rtpmap:0 PCMU/8000 01-08 04:38:40.453: DEBUG/libpjsip(598): a=rtpmap:8 PCMA/8000 01-08 04:38:40.453: DEBUG/libpjsip(598): a=rtpmap:102 speex/8000 01-08 04:38:40.453: DEBUG/libpjsip(598): a= 01-08 04:38:40.463: INFO/libpjsip(598): 04:38:40.472 pjsua_wrap.cpp Call 0 state changed to CALLING 01-08 04:38:40.583: DEBUG/libpjsip(598): 04:38:40.586 pjsua_core.c RX 337 bytes Response msg 100/INVITE/cseq=20133 (rdata0x34bf2c) from UDP 212.27.52.5:5060: 01-08 04:38:40.583: DEBUG/libpjsip(598): SIP/2.0 100 Trying 01-08 04:38:40.583: DEBUG/libpjsip(598): Call-ID: Oe-.NHAmoQNc44Njh6sMgx7LfUwypFAX 01-08 04:38:40.583: DEBUG/libpjsip(598): CSeq: 20133 INVITE 01-08 04:38:40.583: DEBUG/libpjsip(598): From: <sip:09xxxxxxxx at freephonie.net>;tag=4z92T2TUU-15F1BmZ.9t-cOOV-x9FlkJ 01-08 04:38:40.583: DEBUG/libpjsip(598): To: <sip:04xxxxxxxx at freephonie.net> 01-08 04:38:40.583: DEBUG/libpjsip(598): Via: SIP/2.0/UDP 83.206.185.81:1113;received=83.206.185.81;rport=1113;branch=z9hG4bKPjGMc7IBRp1fZEpPMmLu1fko8JALVv69na 01-08 04:38:40.583: DEBUG/libpjsip(598): Content-Length: 0 01-08 04:38:40.583: DEBUG/libpjsip(598): 01-08 04:38:40.583: DEBUG/libpjsip(598): --end msg-- 01-08 04:38:40.583: DEBUG/libpjsip(598): 04:38:40.593 pjsua_core.c RX 700 bytes Response msg 407/INVITE/cseq=20133 (rdata0x34bf2c) from UDP 212.27.52.5:5060: 01-08 04:38:40.583: DEBUG/libpjsip(598): SIP/2.0 407 authentication required 01-08 04:38:40.583: DEBUG/libpjsip(598): Call-ID: Oe-.NHAmoQNc44Njh6sMgx7LfUwypFAX 01-08 04:38:40.583: DEBUG/libpjsip(598): Contact: <sip:04xxxxxxxx at 172.17.20.241:5062;user=phone> 01-08 04:38:40.583: DEBUG/libpjsip(598): CSeq: 20133 INVITE 01-08 04:38:40.583: DEBUG/libpjsip(598): From: <sip:09xxxxxxxx at freephonie.net>;tag=4z92T2TUU-15F1BmZ.9t-cOOV-x9FlkJ 01-08 04:38:40.583: DEBUG/libpjsip(598): Proxy-Authenticate: Digest realm="freephonie.net",nonce="0107265e4dbc26c2104b331dxxxxxxxx",opaque="0105xxxxxxxx",stale=false,algorithm=MD5 01-08 04:38:40.583: DEBUG/libpjsip(598): Record-Route: <sip:212.27.52.5:5060;transport=udp;lr> 01-08 04:38:40.583: DEBUG/libpjsip(598): To: <sip:04xxxxxxxx at freephonie.net>;tag=00-08174-01072763-237b79893 01-08 04:38:40.583: DEBUG/libpjsip(598): Via: SIP/2.0/UDP 83.206.185.81:1113;received=83.206.185.81;rport=1113;branch=z9hG4bKPjGMc7IBRp1fZEpPMmLu1fko8JALVv69na 01-08 04:38:40.583: DEBUG/libpjsip(598): Allow: UPDATE,REFER,INFO 01-08 04:38:40.583: DEBUG/libpjsip(598): Server: Cirpack/v4.42q (gw_sip) 01-08 04:38:40.583: DEBUG/libpjsip(598): Content-Length: 0 01-08 04:38:40.583: DEBUG/libpjsip(598): 01-08 04:38:40.583: DEBUG/libpjsip(598): --end msg-- 01-08 04:38:40.593: DEBUG/libpjsip(598): 04:38:40.595 pjsua_core.c TX 378 bytes Request msg ACK/cseq=20133 (tdta0x3ee4d8) to UDP 212.27.52.5:5060: 01-08 04:38:40.593: DEBUG/libpjsip(598): ACK sip:04xxxxxxxx at freephonie.net SIP/2.0 01-08 04:38:40.593: DEBUG/libpjsip(598): Via: SIP/2.0/UDP 83.206.185.81:1113;rport;branch=z9hG4bKPjGMc7IBRp1fZEpPMmLu1fko8JALVv69na 01-08 04:38:40.593: DEBUG/libpjsip(598): Max-Forwards: 70 01-08 04:38:40.593: DEBUG/libpjsip(598): From: <sip:09xxxxxxxx at freephonie.net>;tag=4z92T2TUU-15F1BmZ.9t-cOOV-x9FlkJ 01-08 04:38:40.593: DEBUG/libpjsip(598): To: sip:04xxxxxxxx at freephonie.net;tag=00-08174-01072763-237b79893 01-08 04:38:40.593: DEBUG/libpjsip(598): Call-ID: Oe-.NHAmoQNc44Njh6sMgx7LfUwypFAX 01-08 04:38:40.593: DEBUG/libpjsip(598): CSeq: 20133 ACK 01-08 04:38:40.593: DEBUG/libpjsip(598): Content-Length: 0 01-08 04:38:40.593: DEBUG/libpjsip(598): 01-08 04:38:40.593: DEBUG/libpjsip(598): --end msg-- 01-08 04:38:40.593: DEBUG/libpjsip(598): 04:38:40.599 tcpc0x3f1c44 TCP client transport created 01-08 04:38:40.603: DEBUG/libpjsip(598): 04:38:40.603 tcpc0x3f1c44 TCP transport 172.20.254.194:50398 is connecting to 212.27.52.5:5060... 01-08 04:38:40.603: DEBUG/libpjsip(598): 04:38:40.606 pjsua_core.c TX 1352 bytes Request msg INVITE/cseq=20134 (tdta0x3dc238) to tcp 212.27.52.5:5060: 01-08 04:38:40.603: DEBUG/libpjsip(598): INVITE sip:04xxxxxxxx at freephonie.net SIP/2.0 01-08 04:38:40.603: DEBUG/libpjsip(598): Via: SIP/2.0/TCP 172.20.254.194:50398;rport;branch=z9hG4bKPjmsIyd.aGHyETsxmkCWLdykws7MpLIgfN 01-08 04:38:40.603: DEBUG/libpjsip(598): Max-Forwards: 70 01-08 04:38:40.603: DEBUG/libpjsip(598): From: <sip:09xxxxxxxx at freephonie.net>;tag=4z92T2TUU-15F1BmZ.9t-cOOV-x9FlkJ 01-08 04:38:40.603: DEBUG/libpjsip(598): To: sip:04xxxxxxxx at freephonie.net 01-08 04:38:40.603: DEBUG/libpjsip(598): Contact: <sip:09xxxxxxxx at 83.206.185.81:1113;transport=UDP> 01-08 04:38:40.603: DEBUG/libpjsip(598): Call-ID: Oe-.NHAmoQNc44Njh6sMgx7LfUwypFAX 01-08 04:38:40.603: DEBUG/libpjsip(598): CSeq: 20134 INVITE 01-08 04:38:40.603: DEBUG/libpjsip(598): Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS 01-08 04:38:40.603: DEBUG/libpjsip(598): Supported: replaces, 100rel, timer, norefersub 01-08 04:38:40.603: DEBUG/libpjsip(598): Session-Expires: 1800 01-08 04:38:40.603: DEBUG/libpjsip(598): Min-SE: 90 01-08 04:38:40.603: DEBUG/libpjsip(598): User-Agent: CSipSimple 01-08 04:38:40.603: DEBUG/libpjsip(598): Proxy-Authorization: Digest username="09xxxxxxxx", realm="freephonie.net", nonce="0107265e4dbc26c2104b3xxxxxxxx", uri="sip:04xxxxxxxx at freephonie.net", response="4dd1ed5axxxxxxxx", algorithm=MD5, opaque="0105fxxxxxxxx" 01-08 04:38:40.603: DEBUG/libpjsip(598): Content-Type: application/sdp 01-08 04:38:40.603: DEBUG/libpjsip(598): Content-Length: 465 01-08 04:38:40.603: DEBUG/libpjsip(598): 01-08 04:38:40.603: DEBUG/libpjsip(598): v=0 01-08 04:38:40.603: DEBUG/libpjsip(598): o=- 2525139520 25 01-08 04:38:40.603: INFO/libpjsip(598): 04:38:40.609 pjsua_wrap.cpp Call 0 state changed to CALLING 01-08 04:39:12.603: DEBUG/libpjsip(598): Stop ringback 01-08 04:39:12.603: INFO/libpjsip(598): 04:39:12.611 pjsua_wrap.cpp Call 0 is DISCONNECTED [reason=408 (Request Timeout)] 01-08 04:39:13.643: DEBUG/libpjsip(598): 04:39:13.643 pjsua_media.c Closing sound device after idle for 1 seconds On 10/09/2010 13:19, Benny Prijono wrote: > It should fall back to UDP. Check for errors in the log. > > Best regards, > Benny > > > > > 2010/9/9 R?gis Montoya<r3gis.3r at gmail.com>: >> Hi, >> >> I've detected a strange behavior with UDP size thresdhold and it's falling >> back to UDP if TCP fails. >> >> I'm using 1.8-trunk but absolutely not sure since from version it was >> introduce, or if it is really a regression or if it's a bad configuration I >> did. >> >> The use case : >> * A sip server that doesn't support TCP. >> * I try to make a call with an invite larger than 1300. >> * It tries first with TCP (=> ok that's fine I have a TCP transport enabled >> for another account and this account is not attached to the UDP tranport, so >> why not...). >> * As expected TCP fails. >> * -> But... it doesn't retry with UDP. >> >> I'm well aware of the fact, if I want to ensure that it doesn't try with >> TCP, I can attach my account to my UDP transport or I can also use the >> PJSIP_DONT_SWITCH_TO_TCP flag. >> But I think that not falling back to UDP is a bug. At least regarding this >> ticket : http://trac.pjsip.org/repos/ticket/831 ("Also when the TCP >> connection fails, the endpoint should revert back to using UDP."). >> >> Is there something I miss or it is a bug? >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
