Telephone between 2 sites using SIP Trunk

[kecubung dalam bacan]

Telephone between 2 sites using SIP Trunk


Hi, colleagues - all in this forum.
Can you help me with the problem I'm having?
Problems:
1. SIP Trunk connection is appropriate
2. But I can't chat, aka no sound appears between IP phones on the 2 sites. On the ip phone, a Conn notification has appeared, but yes, there is no sound either.

Here I attach the SSA capture and Log Monitor:


333759mS SIP Rx: UDP 10.140.0.253:5060 -> 10.210.200.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.210.200.1:5060;rport;branch=z9hG4bKf318122015ded47e7e637613148a7f69
From: <sip:10.210.200.1>;tag=06415fb84424eb3f
Call-ID: 3f65727bf579e9cb12cdda05fb12eb28
CSeq: 487665660 OPTIONS
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.1.2.1.0 build 3
Content-Type: application/sdp
To: <sip:10.140.0.253>;tag=9d690ea25efe0683
Content-Length: 217

v=0
o=UserA 314516031 2002417562 IN IP4 10.140.0.253
s=Session SDP
c=IN IP4 10.140.0.253
t=0 0
m=audio 8000 RTP/AVP 8 0 18
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no

 

[IPGuru]

No Speech means a data routing issue.

I would guess that the vlan with the IP handsets does not have a valid route out to the internet for the direct speech packets
solutions :-
0) IF both sites are IP Office don't use SIP use SCN over a proper private ling (VPN or MPLS) this improved operation will soon pay for itself!
1) fix the data routing
2) disable direct media path on the sip trunks (this is not ideal but probably ok in this case)



Do things on the cheap & it will cost you dear

ACSS

 

[kecubung dalam bacan]

Hi IPGuru,

Permission to answer the questions you submitted.
0) Using MPLS
1) Fix the data routing, may I know what it includes?
2) For direct media path the position is disabled.

Thanks.
Regards.