Snom 370 without STUN with NAT + port forwarding: remotely ended calls do not end
Situation:
PePLink Balance 300 router with:
Snom 370 phone:
Voxalot settings:
Making an outbound call, that is remotely ended, is at:
What can be done to optimize transmission of audio and the call ended message, without using STUN? Ad.1 - incorrect call end (using budgetphone via voxalot) No SIP messages arrive around the end of the call Ad.2 - correct call end (using budgetphone without voxalot) Two last messages arrive in the Snom 370 SIP trace: --- Received from udp:83.143.188.165:5060 at 4/8/2010 03:08:33:931 (482 bytes): BYE sip:xxx@92.254.51.247:5060;line=pw2w4upe SIP/2.0 Via: SIP/2.0/UDP 83.143.188.165;branch=z9hG4bKf0e4.d667c1.0 Via: SIP/2.0/UDP 83.143.188.161:5060;rport=5060;received=83.143.188 .161;branch=z9hG4bK1677465537 From: <sip:444@sip1.budgetphone.nl>;tag=127746832 To: "xxx" <sip:xxx@sip1.budgetphone.nl>;tag=za6yn2wbgs Call-ID: 3c2709321e25-hu9l9g1dczt9 CSeq: 3 BYE Max-Forwards: 9 Reason: Q.850 ;cause=16 ;text="Normal call clearing" Content-Length: 0 Sent to udp:83.143.188.165:5060 at 4/8/2010 03:08:33:965 (610 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 83.143.188.165;branch=z9hG4bKf0e4.d667c1.0 Via: SIP/2.0/UDP 83.143.188.161:5060;rport=5060;received=83.143.188 .161;branch=z9hG4bK1677465537 From: <sip:444@sip1.budgetphone.nl>;tag=127746832 To: "xxx" <sip:xxx@sip1.budgetphone.nl>;tag=za6yn2wbgs Call-ID: 3c2709321e25-hu9l9g1dczt9 CSeq: 3 BYE Contact: <sip:xxx@192.168.254.10:5060;line=pw2w4upe>;reg-id=1 User-Agent: snom370/8.2.35 RTP-RxStat: Total_Rx_Pkts=2359,Rx_Pkts=2352,Rx_Pkts_Lost=0,Rem ote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=4705,Tx_Pkts=2352,Remote_Tx_Pkts=234 6 Content-Length: 0 Ad.4 - Audio stops after 30 seconds, when user now ends call at the Snom 370 phone, 2 SIP messages are in the Snom 370 SIP trace: --- Sent to tcp:85.17.212.38:5060 at 4/8/2010 03:20:07:610 (740 bytes): BYE sip:*1*312998-002314NNNNNNNN@85.17.212.38:5061;natp1eu1=yes SIP/2.0 Via: SIP/2.0/TCP 192.168.254.10:5060;branch=z9hG4bK-4c9m6vhrb6fh;rport Route: <sip:85.17.212.38;transport=tcp;r2=on;lr=on;ftag=0 vs0tsvdi5> Route: <sip:85.17.212.38;r2=on;lr=on;ftag=0vs0tsvdi5> From: "yyy" <sip:312998@eu.voxalot.com>;tag=0vs0tsvdi5 To: <sip:04NNNNNNNN@eu.voxalot.com;user=phone>;tag=as3 674506e Call-ID: 3c270bea70e5-2kffi8lzqmf4 CSeq: 3 BYE Max-Forwards: 70 Contact: <sip:312998@192.168.254.10:5060;transport=tcp;line =dg9k2vhe>;reg-id=1 User-Agent: snom370/8.2.35 RTP-RxStat: Total_Rx_Pkts=1562,Rx_Pkts=1562,Rx_Pkts_Lost=0,Rem ote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=4525,Tx_Pkts=2262,Remote_Tx_Pkts=0 Content-Length: 0 Received from tcp:85.17.212.38:5060 at 4/8/2010 03:20:07:700 (439 bytes): SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/TCP 192.168.254.10:5060;received=92.254.51.247;branch= z9hG4bK-4c9m6vhrb6fh;rport=1046 From: "yyy" <sip:312998@eu.voxalot.com>;tag=0vs0tsvdi5 To: <sip:04NNNNNNNN@eu.voxalot.com;user=phone>;tag=as3 674506e Call-ID: 3c270bea70e5-2kffi8lzqmf4 CSeq: 3 BYE User-Agent: voxalot Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 |
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