SIP audio two-way problem

[rb2009]

Hi,

We are testing generic SIP devices and found a problem.

Internal calls from SIP device to Mitel phone devices are working without any problems.
If I call from SIP device to external number, then I have a problem, that external partner can’t hear me, but I can hear him.

Here is our configuration of the SIP Device Capabilities Assignment:


Disable Reliable Provisionable Response: No
Force Sending SDP in initial Invite message: No
Prevent the Use of IP Address 0.0.0.0. in SDP Messages: Yes
Replace System based with Device based In-Call Features: No
Suppress the Use of SDP Inactive Media Streams: Yes
Renegotiate SDP To Enforce Symmetric Codec: No
Repeat SDP Answer If Duplicate Offer Is Received: No
Allow Device to use Multiple Active M-Lines: No

Any idea what we can do, to solve the problem?

Thanks !



2x ICP 3300 Cluster
Software Release 9.0 UR2 (9.0.2.17)
No Voice Mail Networking licenses
No ACD agent licenses

 

[slapin]

Is it possible to force your SIP device to use G711 codec only? Enable "Force Sending SDP in initial Invite message"
Plus would be nice to know if you have DSP modules installed and have compression licenses. Knowing type of external lines also might be helpful. And if you don't mind to name your SIP device or soft phone, that would be great help

 

[IrwinMFletcher]

When you say external, do you mean external SIP trunk, or TDM? Meaning, does the IP streaming terminate on the controller? If it is a TDM trunk, I assume that that set cannot join a conference either?

If that's the case, then it sounds like the SIP device isn't streaming to the correct UDP port. Wireshark traces are the best at debugging this sort of problem.

 

[rb2009]

Hi,

the SIP device is a Siemens Gigaset C475IP phone. I'll try it with G711 codec. I'm not sure, if we have DSP modules and compression licenses, but I'll will check it tomorrow, when I am in the office.

Connection to external PSTN is over the E1 QSIG interconnection to old Ericsson pbx.

This weekend we are going live with Mitel PBX and then the PBX will be connected directly to PSTN over E1 line.

Maybe this problems disappears after connected directly to PSTN.

Thanks!

 

[rb2009]

Hi,

we could solve the problem with Siemens Gigaset SIP devices.

Here is our configuration of the SIP Device Capabilities Assignment witch are working in our case:

Disable Reliable Provisionable Response: No
Force Sending SDP in initial Invite message: No
Prevent the Use of IP Address 0.0.0.0. in SDP Messages:NO
Replace System based with Device based In-Call Features: YES
Suppress the Use of SDP Inactive Media Streams:NO
Renegotiate SDP To Enforce Symmetric Codec: YES
Repeat SDP Answer If Duplicate Offer Is Received: No
Allow Device to use Multiple Active M-Lines: No



2x ICP 3300 Cluster
Software Release 9.0 UR2 (9.0.2.17)
No Voice Mail Networking licenses
No ACD agent licenses