2. Java version:1.7
3. OS type and the version: 64-bit RHEL6.6
4. UA (phone), gateway or other hardware/software involved:
Cisco Call Manager
5. Your problem:
Our legacy SIP server is sending a SIP call to Cisco Call Manager. Cisco Call Manager is sending the bandwidth modifier in the SDP of 200OK. Our Legacy SIP server does not like the bandwidth modifier and sends BYE as soon as it detects the presence of bandwidth modifiers in the SDP.
We are thinking to proxy the call to Cisco Call Manager via BSS.
Question is how to write the dial the plan to strip the bandwidth modifier if present in SDP of 180/183/200/ or ReInvite.
Code: Select all
These are the bandwidth modifier I would like to strip from the SDP coming from the far side.
b=TIAS:64000
b=AS:64
Code: Select all
Here is complete 200 OK
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.100
To: 999999<sip:999999@10.10.10.200:5060>;tag=121441051~691989bc-6db5-4383-b538-ec917c34eb2d-38772551
From: 555555<sip:555555@10.10.10.100:5060>;tag=5852663a79e8
Call-ID: CANTATA21.3a.3832296.600@10.10.10.100
CSeq: 1 INVITE
Date: Fri, 22 Sep 2017 05:42:46 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence, kpml
Supported: replaces
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Server: Cisco-CUCM10.5
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Session-Expires: 3660;refresher=uas
Require: timer
P-Asserted-Identity: "call to Japan" <sip:2532@10.184.254.23>
Contact: <sip:999999@10.10.10.200:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 227
v=0
o=CiscoSystemsCCM-SIP 121441051 1 IN IP4 10.10.10.200
s=SIP Call
c=IN IP4 10.10.10.200
b=TIAS:64000
b=AS:64
t=0 0
m=audio 8640 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15