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Unread 08-04-2010, 01:33 AM   #1
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Default Snom 370 without STUN with NAT + port forwarding: remotely ended calls do not end

PePLink Balance 300 router with:
  • SIP ALG enabled
  • NAT enabled
  • TCP/UDP 5060 forwarded to internal IP of SIP device
  • UDP:10000-11000 forwarded tointernal IP of SIP device
  • Outbound traffic from SIP device routed over a single WAN link

Snom 370 phone:
  • Running firmware version 8.2.35 29735
  • Identity #1 = Voxalot
  • Identity #2 = Budgetphone
  • Identity #1+#2 Symmetric NAT Handling = No
  • Identity #1+#2 have most identical settings (except username/password/etc.)
  • Dynamic RTP port start = 10000
  • Dynamic RTP port stop = 11000

Voxalot settings:
  • Symmetric NAT Handling = Yes
  • Voice Service Provider "Budgetphone", Optimize Audio Path = Yes

Making an outbound call, that is remotely ended, is at:
  1. Snom370, identity #1 "Voxalot" (routed via Budgetphone): not ended
  2. Snom370, identity #2 Budgetphone directly: correctly ended
  3. Nokia E71, identity Voxalot (to Budgetphone): correctly ended (this phone is Wifi connected via a Fonera+ NAT device and then connected to the PePLink router, so a double NAT-ed connection, using "")
  4. Snom370, identity #1 "Voxalot" (routed via 12 voip (betamax)): the audio channel is lost after 30 seconds [no matter whether is used or not].

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: at 4/8/2010 03:08:33:931 (482 bytes):

BYE sip:xxx@;line=pw2w4upe SIP/2.0
Via: SIP/2.0/UDP;branch=z9hG4bKf0e4.d667c1.0
Via: SIP/2.0/UDP;rport=5060;received=83.143.188 .161;branch=z9hG4bK1677465537
From: <>;tag=127746832
To: "xxx" <>;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: at 4/8/2010 03:08:33:965 (610 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP;branch=z9hG4bKf0e4.d667c1.0
Via: SIP/2.0/UDP;rport=5060;received=83.143.188 .161;branch=z9hG4bK1677465537
From: <>;tag=127746832
To: "xxx" <>;tag=za6yn2wbgs
Call-ID: 3c2709321e25-hu9l9g1dczt9
CSeq: 3 BYE
Contact: <sip:xxx@;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: at 4/8/2010 03:20:07:610 (740 bytes):

BYE sip:*1*312998-002314NNNNNNNN@;natp1eu1=yes SIP/2.0
Via: SIP/2.0/TCP;branch=z9hG4bK-4c9m6vhrb6fh;rport
Route: <sip:;transport=tcp;r2=on;lr=on;ftag=0 vs0tsvdi5>
Route: <sip:;r2=on;lr=on;ftag=0vs0tsvdi5>
From: "yyy" <>;tag=0vs0tsvdi5
To: <;user=phone>;tag=as3 674506e
Call-ID: 3c270bea70e5-2kffi8lzqmf4
CSeq: 3 BYE
Max-Forwards: 70
Contact: <sip:312998@;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: 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;received=;branch= z9hG4bK-4c9m6vhrb6fh;rport=1046
From: "yyy" <>;tag=0vs0tsvdi5
To: <;user=phone>;tag=as3 674506e
Call-ID: 3c270bea70e5-2kffi8lzqmf4
CSeq: 3 BYE
User-Agent: voxalot
Content-Length: 0
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