No speech path over SIP Trunk between CS1K and IPO

[nortavaya]

Hello team,

We have configured a SIP Trunk between CS1K and IPO, all works fine

We can call from each site to another without any issue

Only one problem, in the CS1K side we have two sites (main and GR redundancy), users of the main site can call IPO users with no issue, however users from GR site (CS1K) can call IPO users but with no speech path

What can be the possibles causes of this issue ?

Can we suspect their network (Firewall, NAT...etc) ?

Thank you in advance

 

[Wanebo]

Lot's of possible causes. Have you done a traceSM to see what is being negotiated between the two sides? If the call is establishing but has no speech path, over half the time that's a codec mismatch. Make sure that both ends have codec sets that coincide. Call setup is via SIP signaling but RTP is dependent upon the media negotiated. Firewall? Possible but unlikely. NAT? Again, possible but even more unlikely since the call establishes.

 

[nortavaya]

Thank you Wanebo for your reply, please note that I have build this Trunk without Session Manager but with NRS, so no traceSM command

But it works fine for the IP Phone registered in the main site of CS1K, only the IP Phone that are in GR site and also they are registered on the same SigServer in CS1K
So I don't believe that can be codec mismatch, because all phones of CS1K (main and GR)are registered to the same SigServer

There is a secondary SS, but it works only if the primary goes down

I will check again on the NRS conf...

Thank you

 

[AvayaHE]

DID you Check VOICE CODEC and Paket sizes are similare ?

 

[KCFLHRC]

Do you have VGW Channels configured at the SMG?

 

[nortavaya]

Thanks for the reply,yes I have the VGW ressources configured on the GR site (through DSP daughterboard on the MGCs)
For the codec, the same is used G711 for both site

 

[bignose21]

The users at the GR site IP or TDM would be the first question?

As you mention it could obviously be network related Firewall/ACL etc. just the the reason for the first question is are you looking at the the DSP cards in the GR site if phones are TDM, or the IP Phone VLAN?

The GR site you mention what is the solution, if its SMG then your Media Gateway and IP phones would be registered at the main site meaning the only SIP trunk involved in the call is the one you already confirmed working for the main site phones. Or is it a separate CS1000 stand alone system that you refer to as GR site?

 

[nortavaya]

Thanks bignose21, the users on the GR site are all IP Phones

The CS1000 is standalone system with two remote MGC on the GR site registered to the main site (Primary Call Server)

The GR site contain only Secondary Call Server (in inactive state) and follower Sig Server

All IP Phones on both main and GR site are registered to the main CS and SS

 

[bignose21]

Well then the SIP trunk etc. is the same as the working calls, and as the call sets up just with no speech the routing etc. is good.

So look at the SIP make sure good codec is negotiated as this can be set against the MGC card in elephant manager. If that looks good then I think you need to look at the network path, do a traceroute from the Phone VLAN and DSP TLAN, look along the path for ACL's or Firewalls, are they counting up any blocked packets etc.

If there is NAT between the two then look at the SIP make sure you are seeing the NAT'd address. I had an issue where the far end was sending the internal IP address, which was supposed to NAT translate the local IP's in the SIP messages in their FW, it wasn't so I was trying to setup a speech path to their internal LAN addresses, and of course no speech.

 

[nortavaya]

Thank you, I checked for the codecs and they are configured good

I will check for the network path, I read some documents talking about SIP ALG that should be enabled on the customer's Firewall and relating to speech path issue

 

[nortavaya]

Hi, I found also some IP Phones on the main site having the same issue with speech path, however other IP Phones on the same site works
When I compared through LD 117 stip tn

I found difference only on the Firmware version of the IP Phone, can this be the problem ?

If else, i will update them

which command I can force Fw update on the IP Phone ?

Thank you

 

[bignose21]

on sig server where set is registered

umsUpgradeAll

you can keep checking where your at with isetShow

 

[nortavaya]

The problem is not with Firmware, I have nom some IP phones with new Firmware works and other no (dead air)

Also some IP phones with old firmware works and other no..!!!!

I don't understand what happens

The IP Phones are registering to the same server, same site...etc

which difference can be with this phones to having issue ??

 

[bignose21]

Data ports setup the same VLAN's, Speed/Duplex, QOS, port security (dot1x), etc. Phones setup the same (full DHCP or is there settings in locally in the phones), if full DHCP have you tried factory defaulting one of the bad sets?

 

[bignose21]

We are currently implementing port security (dot1x) in one of our sites and sets have no speach path as the traffic is blocked as the port security doesn't like seeing same MAC on Data and Voice VLAN. If you manually set the Voice VLAN ID in the phone and turn off the Data VLAN it gets around this. If using Cisco data switches if you do a "show mac address-table interface x/x/x" do you see any differences between a good and bad set?

 

[nortavaya]

Thanks for your quick reply, yes they are on the same vlan, same DHCP but with partial configuration (DHCP delivers only IP address, subnet mask and default GW), we set manually the other settings (S1, S2, port...) on the phones

Maybe, I will try the factory default of bad sets

Thank you

 

[nortavaya]

Hello, I did factory reset of the ip phone that having the issue (dead air though sip trunk)but still the same issue

I have use his profile through virtual office whithin another IP Phone (that was working) it works correcly (no issue with dead air over sip trunk)

I have connect the bad set to the correct port (on the Switch) still not working

I think the issue is relating to the phone Physicalay

The Firmware is the same...

Do you have other idea...??

Thank you

 

[bignose21]

Only other thing is setup port mirror to PC port of the phone and use wireshark to collect output during call attempt, you might see the issue in the log.

 

[nortavaya]

Thank you; you know how to enable mirroring on the IP Phone 1120 for example ?

Thank you

 

[bignose21]

In later firmware think it's in the network diagnostics (double press the services key)

Old method was

Mute
Up
Down
Up
Down
Up
Mute
7
Release

If I remember rightly