One Way Audio (But Only Internal Cals Between 2 Sites)

[Turtlepuke]

I'm curious what your troubleshooting process would be on this....

2 remote warehouses 50 miles apart both register back to the same CM. Both sites have all SIP phones.
Sites can make and receive all calls as normal EXCEPT when site 1 calls site 2, site 2 is the only one who can hear the other person.

If site 2 calls site 1 there is no problem with either party. The sites have their own CM locations and IP Network Region. They are both using the same codec set to talk to the other site (G.711.)

I have not been able to get a CM trace yet.

My question is, in this instance what would be more valuable? A CM Trace or a trace of the Session Manager since these are SIP phones?
I'm new to SIP. Does a traceCM show internal calls between two SIP phones?

Would you suspect that this issue is likely a carrier circuit issue with one of the two warehouses?

 

[kyle555]

SM trace. You're looking for the SDP in the Invites/200OKs/ACKs

They're probably sending audio direct to one another and there's a missing IP route or RTP ports not opened. The RTP ports used are defined in the settings file for SIP phones

You know both phones can talk to the SM in the core. If you make both regions not shuffle and force them to pin to a G450 in the core, it'll work and prove the problem to be connectivity between the two phone's subnets.

 

[Turtlepuke]

Thanks Kyle! Is this the setting you're referring to?

 

[kyle555]

Yeah. If they're both in different NRs, and you have Inter-Region to No, they'll need a DSP and likely be forced to a gateway in a NR they can both reach.

Are there local gateways on site? If so, it might need a little more rejigging to make the RTP flow thru the core, but even at that, if you list trace and there's a GW involved, you'll see packet performance every 10 seconds in the trace. A * for packet loss means all packets were lost.

 

[jimbojimbo]

Unfortunately SIP and H.323 phones have completely different feature sets. If you have any H.323 phones at the sites you can run a traceroute in CM from one phone to the other. If they are all SIP you have a few options.

First off, if they are not on an isolated subnet and are on the same subnet as the PCs simply run a tracert from a PC at each site to the other. Look for dropped traffic, loops, and asynchronus routing.

A really great tool is Avaya's SLAmon. There is an agent embedded in the phone which the SLAmon can use to perform tasks such as connecting between devices at different sites, and testing RTP traffic flow. Provides a grid of network status. I strongly recommend this tool for any multi-site system unless you have something else you are using such as Prognosis.

I am also "assuming" you have correlated the Network-Regions with CM or have gone through each and every location created in System Manager to verify the subnets were not duplicated. If you use correlation then the location data is automatically built from CM.

You can remove all inter-region connectivity between the sites and simply "anchor" the call to a media gateway or media server in another region (as suggested above). If your audio is fixed then you might want to beat on your network team until they fix the routing/firewall issues.

 

[Turtlepuke]

Sounds good. Thanks!!!