SIP ALG not rewriting SIP headers

I enable SIP ALG but Peplink Balance 710 is not rewriting SIP headers. Anyone else have the same problem and any suggestion to solve?

Thanks

Do you have more information for this ?

  1. What is the firmware version running for your device ?

  2. WAN Routing Mode ? NAT or IP forwarding ?

  3. SIP ports info ?

  4. Direction for the traffics ? LAN to WAN ?

  5. New deployment ? Initially the SIP connection working fine ?

  6. Any sample packet capture logs can share here ?

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Hi,
What is the firmware version running for your device Peplink Balance 710 Firmware 6.3.1 build 1631

WAN Routing Mode ? NAT

SIP ports info ? UDP and TCP 5060,11000

Direction for the traffics ? WAN

New deployment ? Yes but SIP ALG do not translates the IP address inside the SIP packet during NAT.

Any sample packet capture logs can share here ?

Internet Protocol Version 4, Src: 118.69.126.28, Dst: 192.168.1.244
Transmission Control Protocol, Src Port: 5060, Dst Port: 52464, Seq: 3416, Ack: 3565, Len: 533
Session Initiation Protocol (NOTIFY)
Request-Line: NOTIFY sip:[email protected]:42644;rinstance=0aed3e1047c5ee7c;transport=tcp SIP/2.0
Message Header
Via: SIP/2.0/TCP 172.16.0.196;branch=z9hG4bK1129.866350a2000000000000000000000000.0
Transport: TCP
Sent-by Address: 172.16.0.196
Branch: z9hG4bK1129.866350a2000000000000000000000000.0
To: sip:[email protected];tag=11b6a573
SIP to address: sip:[email protected]
SIP to tag: 11b6a573
From: sip:[email protected];tag=2c4a592da76857e5564d011de26e7b3a-6049
SIP from address: sip:[email protected]
SIP from tag: 2c4a592da76857e5564d011de26e7b3a-6049
CSeq: 2 NOTIFY
Call-ID: 88253YTE0YTgxYmRjY2ZjOWYwNGE0NWRiNzE0N2ViMzNiZGQ
Content-Length: 0
Max-Forwards: 70
Event: message-summary
Contact: sip:172.16.0.196:5060;transport=tcp
Contact URI: sip:172.16.0.196:5060;transport=tcp
Subscription-State: active;expires=299

The IP Address in SIP Message header not change to NAT Public IP so the client can not connect

We do support UDP SIP passthrough only at the moment. TCP SIP passthrough should be a feature request and we will look into this.

Thanks.

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