Sure,
Update this thread if it works for you.
--[[
Description:
pTime attribute is present in SDP when the call is between SIP Phone to non-SIP device (SCCP phone/H323 gateway/MGCP gateway).
Sip Phone to SIP trunk/Phone CUCM would not include a pTime attribute in the SDP sent to the phone.
This script will add a pTime value in the SDP sent to the phone from CUCM when pTime is not present in out bound SDP for G729 codec only.
Script Parameters:
* The following needs to added while adding/uploading this script to SIP Normalization Script:
Memory Threshold : 100 kilobytes
Lua Instruction Threshold : 2000 instructions
* The following needs to added in Sip profile page associated with the Sip phone:
Parameter Name : G729pTime
Parameter Value : 10, 20, 30, 40, 50, or 60 (milliseconds)
-- * This is mandatory parameter.
-- Configure this parameter and make sure this reflects the same value as service parameter “Preferred G.729 Millisecond Packet Size”
Logic:
Check if script parameter is configured, if not then no further operations needs to be performed
Check if SDP is present in the outbound message, if not then no further operations needs to be performed
Check if pTime value is valid (one among 10, 20, 30, 40, 50, or 60) , if not then no further operations needs to be performed
Get the audio mline from SDP
Check if pTime attribute is already present, if not then no further operations needs to be performed
Check if there are more than one Audio codec, if there are more than one codec we need not add pTime attribute.
Add pTime attribute.
--]]
M = {}
-- Read the Script Parameters
local g729ptimeValue = tonumber(scriptParameters.getValue("G729pTime"))
local function addpTimeAttribute(msg)
-- trace.enable()
if g729ptimeValue then
-- if we have value we are good to go
else
-- since there is no Script Parameter we exit this function
-- trace.format("Script Parameter not configured")
return
end
local sdp = msg:getSdp()
if sdp
then
-- check if the pTime value is valid
if g729ptimeValue == 60 or g729ptimeValue == 50 or g729ptimeValue == 40 or g729ptimeValue == 30 or g729ptimeValue == 20 or g729ptimeValue == 10
then
-- We have got a valid pTime value
else
-- trace.format("Script Parameter ptime is NOT valid")
return
end
-- get audio mline
local audio = sdp:getMediaDescription("audio")
if audio
then
-- check if we already have ptime
-- if yes do not proceed
local ptimeLine = audio:getLine("a=","ptime")
if ptimeLine then
-- trace.format("we have ptime do not porceed")
return
end
local tokenCount = 0
-- split the m=audio lines using space as delimiter
-- for the like below the codecs would start from token 4
-- m=audio 19800 RTP/SAVP 18 101
-- if the number of codecs are more than 2 then skip and
-- do not proceed further
local audiomLine = audio:getLine("m=","audio")
for token in string.gmatch(audiomLine, "[^%s]+") do
tokenCount = tokenCount + 1
if tokenCount > 5
then
-- we got multiple codecs in SDP hence we need not modify it
return
end
end
-- we have 2 codecs now need to make sure one is RFC2833
-- else this a case of multiple audio codecs
-- do not proceed.
local rfc2833_line = audio:getLine("a=","telephone-event")
if rfc2833_line then
-- if the 2nd codec is telephone-event then its ok to add the pTime
else
-- if the 2nd codec is not telephone-event then this could be a multiple codec case
-- so return out
return
end
-- now that we have only one codec with no ptime
-- go ahead and add ptime value.
-- the following api will add the ptime value only if it finds G729
audio = audio:insertLineAfter("a=", "G729", "a=ptime:" .. g729ptimeValue .. "\r\n")
sdp = sdp:modifyMediaDescription("audio", audio)
msg:setSdp(sdp)
end
end
end
M.outbound_INVITE = addpTimeAttribute
M.outbound_ANY_INVITE = addpTimeAttribute
M.outbound_ACK = addpTimeAttribute
return MAfter adding this normalization script, add it to your SIP profile. Make sure to add the relevant parameter in the SIP profile.
