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Capture RTP on CS1K phones

Capture RTP on CS1K phones

Capture RTP on CS1K phones

(OP)
Hello team,

I have issue with some ip phones (1100 series) connected to the CS1K system, when they try to call from CS1K to IP Phones located on IP Office system (through SIP Trunk) the call rings, however no speech path

But some other phones work fine (from CS1K side), I checked everything between good and bad phones (FW, IP, ports, config, TN...etc) all configs are similar, the only thing different is the MAC address

I tried to capture RTP traffic from this phones, but I didn't got enough informations through mirrored port of 1100 phone

Do you have an idea from where I can have a complete RTP traces ? and perhaps what can be the issue ?

Thank you in advance

RE: Capture RTP on CS1K phones

My guess is it could be an issue with the signal server they are registered to. If you have more than 1 signal server in the Node they should load share. It's possible that the phones that don't work are all registered to the same SS, just a guess though. Voice traffic is carried over the TLAN and call set up is carried over the ELAN, so I would start looking at the TLAN Subnet.

RE: Capture RTP on CS1K phones

(OP)
Thank you KCFLHRC for you reply, of corse we have two SS Server, the first one is located on the main site and the second on the GR site, also all IP Phones on the GR site are having this issue (no speech path)

But, on the IP configuration of all IP phones, we have already set S1 as primary SS and S2 as secondary

I will check how I can printout the ip phones registered by SS1 and SS2

RE: Capture RTP on CS1K phones

There is NRS programming that has to be completed properly for the Geo Redundancy to work properly. Although it may sound silly the S1 IP at the Geo site should really be the SS at the Geo site and then NRS redirects it to the primary SS if it is reachable. You really shouldn't configure a S2. Don't shoot the messenger, but that is the way it is supposed to be done.

RE: Capture RTP on CS1K phones

(OP)
Thank you KCFLHRC, so you mean that the NRS configuration can be the issue ? however the call from GR site to the other site through SIP trunk works, it is just issue with speech path

RE: Capture RTP on CS1K phones

in geo redundancy you are supposed to config S1 as the local SMG Node IP and S2 as the Main Site Node IP

The reason is like KCFLHRC say that it first registers with the S1 at the SMG and is then forwarded to the main site. If the S1 fails as the sig server is goosed at the SMG site for example, then after the number of configured time outs it will register direct to the main site using the S2.

RE: Capture RTP on CS1K phones

log in to the SMG callserver and in LD 117 do a "stat ss" or "ecnt zone" and you should not see any registered sets. Or in the Sig Server at the SMG site do an "isetShow" which should also show no registered sets.

For example this is one of our SMGs

LD 117
=> ecnt zone 106
Zone: 106
Number of Registered Ethersets : 0
Number of Unregistered Ethersets : 5


And then the main site

=> ecnt zone 106
Zone: 106
Number of Registered Ethersets : 5
Number of Unregistered Ethersets : 0


=>

RE: Capture RTP on CS1K phones

Actually the S2 shouldn't really be needed at all because S1 is the GEO Site and NRS routes it to the primary site. Then if the GEO site cant redirect to the main then it will automatically register to the GEO site. Like I said, NRS has to be set up properly for it to work properly.

RE: Capture RTP on CS1K phones

And if the geo Sig server or call server are down for any reason sets won't register unless you config the s2

RE: Capture RTP on CS1K phones

(OP)
Hello, I don't think this issue is relating to the Geo configuration, because all ip sets are registered on the main Sig Server

To confirm this, I did a test by bringing IP Phone that was working fine on main site and when I connect it on the GR site network, we got the issue of dead air when calling over SIP Trunk

I suspect issue with network rules (Firewall, router...etc) configured between the GR site and IPO site, may they are blocking RTP streams

Because all GR site are having this issue of dead air, once we bring one phone from GR site and put it on the main site network, it works fine

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