Genesys SIP Codec Filtering

CODEC Filtering

SIP as such supports wide variety of CODEC - pcmu,pcma,g711,g726,g729,g722,gsm,amr amp-wb h263...

while you may have your own codec preference which you would want to enforce - 

Also you may want prioritize your response over a local preference than a remote preference -(local preference means option configure in Genesys would take preference while Remote means SM or the preference from the gateway will be used.

if you would want to set a preference over pcmu over puma and in a INVITE which contain both PCMU 0 and PCMA 18 in it, if you want to communicate over just PCMU 0 , the you could set your config as follows :

In SIP Server - set the audio-codecs- telephone-event, PCMU,PCMA,G711,G729.G722....(list of supported protocols) and set sip-enable-sdp-codec-filter to be true. 

following this in MPC - 

set codecpref to be local and codec option - pcmu,pcma,g711,g726,g729,g722

and also set answerwithonecodec to be 1 - 

by this way you could get the response with single codec, set the preference to be local preference and also limit the distribution of rftpmap message that is sent on the re-invite message to gateway back. 

so this would be like, 

if the invite comes with PCMU and PCMA - (per the above preference and config set) we would respond back with PCMU.(PCMU will be negotiated) 

if invite comes with PCMU only then PCMU only will be negotiated. 

If invite comes with PCMA only then PCMA only will be negotiated.

SIP PreConsiderations

                                                      Few Things to Consider

• Have both TCP and UDP ports opened(you can make it work , later you can decide the protocol standards)
• UDP is anyways essential RTP goes over UDP even if your organization standard is TCP
• Don't install RM and SIP on the same server ( SIP and RM will fight for the port if default configurations are used and it would cause trouble) 
• If you would like to have RM and SIP in the same server configure them on different ports carefully(SIP - 5060 and RM 7060 for example) 
• Have Session Manager IP /Gateway details handy. 
• If you are going to host SIP Proxy - have the SRV Records(SIP Proxy needs SRV Records as a part of configuration ) I will write a post on setting up SRV and Proxy details. 
• If you are doing IP takeover for HA - you may need an IP address in same subnet that is available (make sure to have it blocked for you ) 
• Consideration of the integration points - if you have Pure SIP or Avaya Agents with SIP , VHT, or any other external IVR application to SIP. SIP-GVP is closely coupled and you don't have to worry much. 
• SIP ERP's accountability based upon volume of call. 
• #' of MCP depending upon call volume and future growth prediction and services that are going to be offered. 
• Reporting considerations
• IP address of RM/MCP/SIP servers handy - this will be needed in setting up most of the configurations. Make sure all these servers are able to communicate with each other and no firewall restriction. Antivirus software may cause troubles at times.
• Codec Consideration and support!!!

Thanks 

SIPAR

What YOU can GET here

To get it straight, if you are new to SIP world and you are implementing SIP for the first time then you could find some information on specific road blocks you may hit -
few integration points to external systems and also shortcuts for certain configurations to achieve few feature which you would eventually find with Service Request or through Google.  This being said - I m going provide as much details as I could ! Common issue/Config Options/Shortcuts and few tips. With this said Shortly will start with my first post.
As always if you find me wrong/incorrect/ or easy way of doing what I have suggested please share them - it will be a great learning for me and for the rest.

Thanks
SIPAR