SIP Server - Limiting RTP Port Range / Codec Priority

[JTL]

I've been scratching my head with this one, as well as reading through all of the documentation for SIP Server... but I can't find exactly the details I'm after.

In a contact centre environment which is primarily used for OCS-based Outbound dialling, using IWS and Agent Scripting, I'm looking for the best way(s) to:

a) Limit the RTP Port range used
b) prioritise negotiation based on G711 rather than G729 (whilst keeping both configured)

I have found, in the IWS documentation, details on how to configure the SIPEndpoint (in the provided XML file) to do both of these things... however, since the initial port also seems to be in the original INVITE, initiated from SIP Server without input from IWS, I am concerned that this approach isn't quite enough.

However, I can't see how to accomplish either of these using a SIP Server configuration option...

So should I be configuring something on the Agent Login / Extension or even on the VTD or Trunk Group DN objects?

[genesysguru]

I assume you are using OCS in predictive dialling mode? Are you using Media Server for CPA/CPD?

The RTP port and codecs offered / preference will be sent in the SDP and negotiated between the two endpoints. A re-INVITE can occur when the endpoint changes. Hence the RTP port range and codecs should be configured on Genesys Media server or IWS SIP endpoint.

There are also some SIP server options worth reviewing:

audio-codecs
sip-enable-sdp-codec-filter

[JTL]

[quote author=genesysguru link=topic=7790.msg33781#msg33781 date=1367163301]
I assume you are using OCS in predictive dialling mode? Are you using Media Server for CPA/CPD?

The RTP port and codecs offered / preference will be sent in the SDP and negotiated between the two endpoints. A re-INVITE can occur when the endpoint changes. Hence the RTP port range and codecs should be configured on Genesys Media server or IWS SIP endpoint.

There are also some SIP server options worth reviewing:

audio-codecs
sip-enable-sdp-codec-filter
[/quote]

I did review the audio-codecs options and the filter codecs one too. Oddly, the environment was NOT configured with G711 in the audio-codecs option, even though Wireshark traces showed it was the main codec being used - so I assume it was perhaps because the sip-enable-sdp-codec-filter was set to 'true' or 'enable' or whatever it was, and therefore it was using the DN-level configuration. Audio-codecs wasn't set at DN-level either - but perhaps expecting it to be present, it was therefore taking on the 'default' configuration which includes everything? I know that the DN-level config should just be a subset of the SIP Server settings, but I assume this is why it was working anyway.

We aren't using any CPD - just a straight Progressive or Preview dial, and leaving messages on answering machines

[cavagnaro]

JTL don't mix the questiions, focus on the one you asked on the topic and then open a new one for IWS

[genesysguru]

Cav - you are a grumpy old bugger at times! 

I was just about to create to new post for JTL with his IWS question but I see that he has amended his previous post. It would have been quicker to do this for him (as I was about to) rather than bitching! JTL - please create as a new post with your prevous IWS question.

This forum is about give and take. On this occasion JTL seems to have posted something useful potentially in the future to all members.

[cavagnaro]

Yes Greg,
I was going to but forgot I was not grumpy but asking him not to mix posts as next guy with similar issue will find information mixed and not clear...just that

[Kubig]

It is necessary to check all participants codec configuration (SIP server, MGW, IWS SIP Endpoint, MediaServer, etc). On Genesys site you have to set on several level of objects, but I would suggest to you configure it on SIP server, SIP endpoint and MediaServer lvl - not on DN level and so on.

[JTL]

[quote author=Kubig link=topic=7790.msg33796#msg33796 date=1367223868]
It is necessary to check all participants codec configuration (SIP server, MGW, IWS SIP Endpoint, MediaServer, etc). On Genesys site you have to set on several level of objects, but I would suggest to you configure it on SIP server, SIP endpoint and MediaServer lvl - not on DN level and so on.
[/quote]

Thanks Kubig. I will make the changes on IWS and re-test.

Thanks also genesysguru... and Happy Birthday?

[genesysguru]

Thanks - yes another year older today!

[cavagnaro]

Where is the party???

[JTL]

BTW, for what it is worth, I had a play with IWS and ClickOnce today... careful not to break the existing setup, I managed to create a test version to deploy, which contained the items we usually use - e.g. some custom ring tones, a corporate directory XML, SipEndPoint, GAS, WebModule and some other bits.

In this version I modified the sip config to limit to specific ports (16xxx to 32xxx) which are the ones preferred on the QoS settings on the network / WAN.

I also added the option for honor_first_codec='1' as we do see random 1-way speech which could well be codec negotiation related.

I managed to deploy this to my test PC and it appears to be working...

I couldn't begin to roll out to contact centres since we were suffering random issues with slow delivery of Agent Scripts - so it is unwise to "play" whilst the environment is unstable - but I may begin rollout to a select handful of users for broader testing later this week.

Thanks again for everyone's help.

BTW - these issues came to the fore when our provider upgraded SBC last weekend (Acme Net-Net) to a later firmware, and suddenly we see an awful lot of disconnections, silent calls, calls which don't disconnect (SIP server not seeing the BYE) and somehow even our outbound CLI is broken as well. A week of issues, you could say...

SBC couldn't even be rolled back, and the HA pair fell over completed with kernel errors too... sounds like a bad firmware

But this is why we also noticed that some traffic is outside of the ranges, and hence why I wanted to make this change. Sorry for going off track, but wanted to give some background.