We are having some problems with calling to external sites. The phone call to that site goes through Voice Gateway in Italy. We've checked IP routing, Call Manager configurations. Also we've checked that VG configurations it all seem ok. Nothing is blocking voice packets or UDP traffic. Last thing we've done is port span where our Cisco IP Phone (7941G) is connected with Wireshark and you did not see any voice packets coming out of that port but coming in voice packets there is no problem.
The problem is we (where call is originated) can hear other side but they cannot hear us. Call registration establishment is ok.
Call manager version is 4.3.
Please, help me out.
I will provide you with additional information if you need.
One way routing is 99.999% result of IP routing issues or firewal blocking the RTP ports.
Is there a firewall or ACL in between?
Have you tried ping and trace route between the voice subnets?
Yes, I've checked ACL on all interfaces between switches and routers it's all ok. Call goes through WAN connection no firewalls are involved. Nothing is blocking RTP packets. Yes, I thought that there might be routing problem but connection between phone and registration server goes normal problem only with converation.
Call registration messages are sent to your registration server whereas RTP traffic is usually directly between the end devices. You will want to verify your IP routing between the 2 phones or the remote gateway if it is going out to the PSTN via a digital/analog line.
Yes, I know that, let me explain network topology. IP phone located on our network and second end device is some cell phone of foreign country. When someone calls it goes from our local network to WAN router then to Voice gateway located in Italy. I did trace of route of packets it ok, but when end devices get connected one side calling side cannot hear anything. I've done some tests and result is RTP packets did not reach even WAN router after connection is established. There is no any ACL between them.
I'm a little bit confused with situation.
Which side cannot hear the other side? Can you ping from the IP phones subnet to the media IP address of the voice gateway in Italy? If you are losing RTP packets somewhere, you can narrow it down to a certain hop using packet captures. That way you can narrow it down to which router is not routing the RTP packets properly.
If I call for example, I can hear voice of second side but second cannot hear me. Per us investigation RTP packets comes to LAN from external site, but our packets do not reach them.
Could be the problem some codec issue or some mismatch of some settings of registration server or voice gateway? Because even Cisco IP Phone do not send RTP packets. I've checked it with port spanning with Wireshark.
Can you attach the Wireshark trace for a call? The phone should be sending RTP packets unless it doesn't have a route to the destination network. Does the phone have a defaqult gateway configured or assigned via DHCP?
I actually already have done it. IP phones get their IP by DHCP. All other calls are ok. And default gateway configured properly. The problem is after call is established our phones do not send RTP only receives them.
There is file. IP phone addr. is 10.156.28.48 def. gateway is 10.156.28.1 and 10.129.2.166 is addr. of voice gateway or server I really do not know it is on different location. 10.156.20.84 is addr. of our Call manager.
I'm having problems playing both audio streams from both directions. It looks like there's a little bit of buzzing and clicking, but that's it. Are these phones using SRTP or anything like that?
What do you mean? Call flow goes to Call Manager after it establishes connection with remote site through WAN router and IP Phone call flow goes through that router. It is connected to our default gateway. WAN router connected with remote site by satelite connection. IP Routing protocol is EIGRP. Route to remote site learned by EIGRP.
No, only one of them. IP phone on our LAN network is registered only. Second phone might cell phone or land line.
Sorry, I'm right now checking pcap file results from our tests. And I think we are using H.225. Is it possible?
H.225 is just the call signaling used by H.323. I think you may want to open a TAC case if you haven't already as I'm running out of ideas. It seems like you are getting valid RTP streams but they are silent. That means routing and codec negotiation should be good. If you don't have SRTP Allowed checkbox, then you don't have cluster security turned on for CUCM so SRTP is probably not the issue.
I've checked our CallManager (version 4.3) trunk configuration and I there is no such option as SRTP allowed. But as per cisco documentation verion 5.0 has this option.
Yes, you are right there is RTP stream but they are silent. I also have no idea what could be problem.
Brian thanks for your help.
But there is another site of ours, they also have same CallManager 4.3, same Cisco devices between sites, same IP phones, same connection through satelite to remote site but they don't have this kind of issue.
It looks like your network subnet is not published to the remote site.
Please get this checked and publish your local network to the remote site.
What do you mean by not published? Route to remote site is learned by EIGRP and it is ok. Or you mean in CUCM? We have trunk that points to some gateway located in remote site in CUCM configuration.