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SIP Client oneway speach
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IrwinMFletcher
Programmer
Router/firewall. Make sure the UDP port range is open.
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3300 on same subnet as phones and wireless? Trunk calls would mean streaming RTP from the IP of the 3300 to the IP of the SIP device. SIP to internal would mean streaming RTP from the IP of the SIP to IP of the 5312.
I'd tell you a UDP joke but I'm afraid you won't get it. TCP jokes are the best because you always get them.
I'd tell you a UDP joke but I'm afraid you won't get it. TCP jokes are the best because you always get them.
IrwinMFletcher
Programmer
I (wrongly) assumed SIP trunks.
Just so I'm clear, you have a HTC phone running a SIP client registered as a user on the 3300, connecting internally through some wifi bridge. If you call from that phone to another IP set on the internal network, it works fine. But if you call from the HTC out a trunk, you get one way audio. Is that correct?
If so, then it's again, most likely a router issue. In the first scenario, the RTP data is being sent from the IP phone, most likely using port 9000. When you call out a trunk, the RTP stream gets terminated on the controller, which uses a different port range. So I'm guessing your routers are configured to allow one, but not the other.
If it were me, I'd just start sniffing the call. Eliminates the guess-work.
Just so I'm clear, you have a HTC phone running a SIP client registered as a user on the 3300, connecting internally through some wifi bridge. If you call from that phone to another IP set on the internal network, it works fine. But if you call from the HTC out a trunk, you get one way audio. Is that correct?
If so, then it's again, most likely a router issue. In the first scenario, the RTP data is being sent from the IP phone, most likely using port 9000. When you call out a trunk, the RTP stream gets terminated on the controller, which uses a different port range. So I'm guessing your routers are configured to allow one, but not the other.
If it were me, I'd just start sniffing the call. Eliminates the guess-work.
Supernova99
Technical User
Have you got an E2T card installed on the controller?
If so is the default gateway correct?
Also has the HTC handset got the correct default gateway from the wifi connection?
It would need to be either the same as the 3300 controller or routable between the gateways if different.
If all is correct I would as IrwinMFletcher has stated start a wireshark trace and find out what and where the RTP is being blocked.
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If so is the default gateway correct?
Also has the HTC handset got the correct default gateway from the wifi connection?
It would need to be either the same as the 3300 controller or routable between the gateways if different.
If all is correct I would as IrwinMFletcher has stated start a wireshark trace and find out what and where the RTP is being blocked.
Share what you know - Learn what you don't
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good sugestions - Thanx
yes IrwinMFletcher your summery of the setup is correct
my reasoning - if the sip client can register on the Mitel and the internal call to a 5312 works then the gateway, network setup should be fine. If LoopyLou is correct and the RTP communication with SIP is between the IP 5312 and the HTC SIP client, then the gateway is in anyhow out of the picture.(flat network)
The Sip Client and 3300 can eatablish a call thus ip coms is correct and with an external call the RTP is between the HTC and the 3300.
It seems the only solution whithout any guess work would be a wireshark trace.
Will post as soon as done
yes IrwinMFletcher your summery of the setup is correct
my reasoning - if the sip client can register on the Mitel and the internal call to a 5312 works then the gateway, network setup should be fine. If LoopyLou is correct and the RTP communication with SIP is between the IP 5312 and the HTC SIP client, then the gateway is in anyhow out of the picture.(flat network)
The Sip Client and 3300 can eatablish a call thus ip coms is correct and with an external call the RTP is between the HTC and the 3300.
It seems the only solution whithout any guess work would be a wireshark trace.
Will post as soon as done
Supernova99
Technical User
another test you can try first is to setup a SIP softphone (x-lite) and see if you have any issues with that
If it works fine then it must be a wifi router problem
Share what you know - Learn what you don't
If it works fine then it must be a wifi router problem
Share what you know - Learn what you don't
IrwinMFletcher
Programmer
Don't get 'being able to establish a call' and 'streaming' confused. They are different entities and require different things.
SIP devices establish a call over port 5060, a well known, pre-defined port. They use that as the means to determine how they will transmit the data, and they can end up picking something arbitrary. If you have a 'SIP-aware' router, then port 5060 will be open, however the UDP ports that the audio is sent over will not be. You'll have to open those by yourself since they are very application/device specific.
SIP devices establish a call over port 5060, a well known, pre-defined port. They use that as the means to determine how they will transmit the data, and they can end up picking something arbitrary. If you have a 'SIP-aware' router, then port 5060 will be open, however the UDP ports that the audio is sent over will not be. You'll have to open those by yourself since they are very application/device specific.
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