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skb007 Brekeke Guru
Joined: 05 Oct 2015 Posts: 152
Location: USA
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Posted: Tue Sep 26, 2017 3:33 pm Post subject: Strip Bandwidth Modifier From SDP in 200OK |
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1. Brekeke Product Name and Version: Brekeke SIP Server 3.6.2.5
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: |
These are the bandwidth modifier I would like to strip from the SDP coming from the far side.
b=TIAS:64000
b=AS:64
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Code: |
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
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janP Brekeke Master Guru
Joined: 25 Nov 2007 Posts: 336
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Posted: Tue Sep 26, 2017 4:55 pm Post subject: |
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Use Brekeke PBX and send a call through the PBX.
As B2B-UA, Brekeke PBX doesn't pass "b=" lines.
If you want to use Brekeke SIP Server instead of PBX, you need to write a plugin to modify SDP. |
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skb007 Brekeke Guru
Joined: 05 Oct 2015 Posts: 152
Location: USA
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Posted: Mon Oct 02, 2017 12:51 pm Post subject: |
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We do not have license for PBX, we are running on a license for BSS.
Is there any way to use $str.remove function to remove bandwidth modifiers.? |
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Ericcc Brekeke Member
Joined: 21 Apr 2014 Posts: 24
Location: NY, USA
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Posted: Mon Oct 02, 2017 4:43 pm Post subject: |
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DialPlan can pick a content of SDP but it can not modify it.
Using Session plugin will be a solution but you need a coding.
Using Brekeke PBX is the easiest way.
http://www.brekeke.com/downloads/pbx.php |
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