Issue between Softphone and Communicator

[dcs9150]

HI
we are having an issue at a client between softphones and communicators
the client has a main site with extensions at a remote site connected via MPLS they use communicators and softphones at both these sites
There is a third site with an IPO SCN'd to the main site and use softphones only
IPO's are 9.1.1

we are having issues between the softphones and communicators at the remote site

calls between the softphones and communicators work fine between each other at the main site, and between any site and third site with the IPO.

issue is in the MPLS site, communicator to softphone just produces ringing for the communicator and dead air when answered on the softphone
Softphone to communicator one way audio.

it looks like an direct media path issue, that is deticked for the communicators but not an option for the auto created softphones.
it looks like a possible routing or port issue but am told it is OK with no restrictions from their IT

I can see a difference in the traces for the communicators at the remote site

After the return of the 180 Ringing this sequence repeats where it doesn't repeat for the communicators at the main site.


2540835501mS SIP Tx: TCP 192.168.0.252:5060 -> 192.168.2.102:54363
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.2.102:54363;branch=z9hG4bK28f_28cd044a442f6a17-7f598200_I2357
From: <sip:2357@192.168.0.252>;tag=-6a6349c357d970a6-7f598a09_F2357192.168.2.102
Call-ID: 28f_28cd044a5a93d759-7f598b0b_I@192.168.2.102
CSeq: 655 INVITE
Contact: "Jason H" <sip:2339@192.168.0.252:5060;transport=tcp>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,SUBSCRIBE,REGISTER,PUBLISH,UPDATE
P-Asserted-Identity: "Jason H" <sip:2339@192.168.0.252:5060>
Supported: timer,100rel
Server: IP Office 9.1.1.0 build 10
To: <sip:2339@192.168.0.252>;tag=936866c21760f875
Content-Type: application/sdp
Content-Length: 227

v=0
o=UserA 3246789746 3707770418 IN IP4 192.168.2.88
s=Session SDP
c=IN IP4 192.168.2.88
t=0 0
m=audio 61188 RTP/AVP 18 120
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:120 telephone-event/8000
a=fmtp:120 0-15
2540837502mS SIP Tx: TCP 192.168.0.252:5060 -> 192.168.2.102:54363
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.2.102:54363;branch=z9hG4bK28f_28cd044a442f6a17-7f598200_I2357
From: <sip:2357@192.168.0.252>;tag=-6a6349c357d970a6-7f598a09_F2357192.168.2.102
Call-ID: 28f_28cd044a5a93d759-7f598b0b_I@192.168.2.102
CSeq: 655 INVITE
Contact: "Jason H" <sip:2339@192.168.0.252:5060;transport=tcp>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,SUBSCRIBE,REGISTER,PUBLISH,UPDATE
P-Asserted-Identity: "Jason H" <sip:2339@192.168.0.252:5060>
Supported: timer,100rel
Server: IP Office 9.1.1.0 build 10
To: <sip:2339@192.168.0.252>;tag=936866c21760f875
Content-Type: application/sdp
Content-Length: 227

v=0
o=UserA 3246789746 3707770418 IN IP4 192.168.2.88
s=Session SDP
c=IN IP4 192.168.2.88
t=0 0
m=audio 61188 RTP/AVP 18 120
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:120 telephone-event/8000
a=fmtp:120 0-15



Any thoughts or ideas would be appreciated

thanks in advance.

 

[kyle555]

Had a customer with SIP phones at a branch and it turns out the firewall had a SIP application layer gateway with no config that was trying to "help". Their IT guy didn't think he had any restrictions either!

Wireshark the messages back and forth and make sure you see each message on each side and if not, go from there.

 

[dcs9150]

Thanks Kyle,

thinking along the same lines with an item in the firewalls.

i'll let you know what we find

thanks,