andreacadia
asked on
Avaya IP Office SIP Trunk Issue
would like to get your take on a resolution to my SIP trunking dilemma. i am configuring a SIP trunk to a provider with the network architecture as:
PBX LAN 1 -> firewall -> provider network
we are successfully registered and inbound and outbound calling is working. however, audio inbound to the IP office is not working. so in other words, when you call into the IPO you can hear the person but they cannot hear you. after much packet analysis on the provider side and on the IP office side it seems that we have narrowed the issue down to SDP. the provider is seeing SDP information to send audio back to the IP address assigned to LAN 1 (192,168,168.30) of the IPO which is behind the firewall. i need the IPO to send SDP information to let the provider know to send audio back to the IP address of the firewall outside interface (10.179.8.2) so it can then be port forwarded (PAT) back to the LAN 1 IP on the IPO.
192.168.168.30 = IP Office LAN 1 (LAN 2 not in use)
10.179.8.2 = firewall outside interface
10.128.2.40 = provider switch (ITSP domain name)
below is a packet capture on an outbound call that completed showing the SDP info on the IP office side and in red are the IP address parameters that need to be changed to the outside interface of the firewall (10.179.8.2). is this possible? i have never had an issue with this before and have confirmed other SIP trunks i have setup are not affected by the private IP of the IPO LAN 1 showing in the SDP info and all other factors being equal:
any and all help is greatly appreciated.
[color #204A87]4244276mS SIP Tx: UDP 192.168.168.30:5060 -> 10.128.2.40:5060
INVITE sip:xxxxxxx@10.128.2.40 SIP/2.0
Via: SIP/2.0/UDP 192.168.168.30:5060;rport; branch=z9h G4bK63bea0 30317fe3f9 cac9ab31d4 8a8fee
From: "Anonymous" <sip:xxxxxxxxxx@10.128.2.4 0>;tag=66d b1c96bc5af 4d6
To: <sip:xxxxxxx@10.128.2.40>
Call-ID: 375a90c5a765f49b42085cd41f 8ac677
CSeq: 1609181771 INVITE
Contact: "Anonymous" <sip:xxxxxxxxxx@192.168.16 8.30:5060; transport= udp>
Authorization: Digest username="xxxxxxxxxx",real m="10.128. 2.40",nonc e="a891416 00011",res ponse="551 15662b590b 77777ca56b 79e8ae97a" ,uri="sip: xxxxxxx@10 .128.2.40" ,algorithm =MD5,qop=a uth,nc=000 00007,cnon ce="086d33 1e52366b1f 4ca8"
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS, BYE,INFO,N OTIFY,UPDA TE
Supported: timer
User-Agent: IP Office 9.1.0.0 build 437
Content-Type: application/sdp
Content-Length: 301
v=0
o=UserA 3773082749 329259799 IN IP4 192.168.168.30
s=Session SDP
c=IN IP4 192.168.168.30
t=0 0
m=audio 8000 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15[/color]
PBX LAN 1 -> firewall -> provider network
we are successfully registered and inbound and outbound calling is working. however, audio inbound to the IP office is not working. so in other words, when you call into the IPO you can hear the person but they cannot hear you. after much packet analysis on the provider side and on the IP office side it seems that we have narrowed the issue down to SDP. the provider is seeing SDP information to send audio back to the IP address assigned to LAN 1 (192,168,168.30) of the IPO which is behind the firewall. i need the IPO to send SDP information to let the provider know to send audio back to the IP address of the firewall outside interface (10.179.8.2) so it can then be port forwarded (PAT) back to the LAN 1 IP on the IPO.
192.168.168.30 = IP Office LAN 1 (LAN 2 not in use)
10.179.8.2 = firewall outside interface
10.128.2.40 = provider switch (ITSP domain name)
below is a packet capture on an outbound call that completed showing the SDP info on the IP office side and in red are the IP address parameters that need to be changed to the outside interface of the firewall (10.179.8.2). is this possible? i have never had an issue with this before and have confirmed other SIP trunks i have setup are not affected by the private IP of the IPO LAN 1 showing in the SDP info and all other factors being equal:
any and all help is greatly appreciated.
[color #204A87]4244276mS SIP Tx: UDP 192.168.168.30:5060 -> 10.128.2.40:5060
INVITE sip:xxxxxxx@10.128.2.40 SIP/2.0
Via: SIP/2.0/UDP 192.168.168.30:5060;rport;
From: "Anonymous" <sip:xxxxxxxxxx@10.128.2.4
To: <sip:xxxxxxx@10.128.2.40>
Call-ID: 375a90c5a765f49b42085cd41f
CSeq: 1609181771 INVITE
Contact: "Anonymous" <sip:xxxxxxxxxx@192.168.16
Authorization: Digest username="xxxxxxxxxx",real
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,
Supported: timer
User-Agent: IP Office 9.1.0.0 build 437
Content-Type: application/sdp
Content-Length: 301
v=0
o=UserA 3773082749 329259799 IN IP4 192.168.168.30
s=Session SDP
c=IN IP4 192.168.168.30
t=0 0
m=audio 8000 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15[/color]
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