No audio in a direction between SIP Phone and UA phone

[jpissot]

Hi,

I have a OXE R9.0 H1.301.28B with two appliance. There are a Crystal Shelf with INTIPB and a MG shelf with GD-2.

Appliances, INTIPB, GD and ipphone are in the domain 1. Intra = without compression, extra = default compression.


I have 9 SIP phone (Siemens Gigaset A580).
They can call exterior. In internal, they can can call ipphone and the UA and Z phones behind the MG shelf.

By cons, when they calls phones connected to the Crystal Shelf, they don't hear their interlocutor. But their interlocutors hears them.

The SIP phone are not on the same LAN that CS, INTIPB, GD. I try also with a lan @, the problem is similary.

I don't understand knowing the GD and INTIPB are affected to the same domain.


Thanks for your helps.

[Konstantinos.E]

hi

if you have all equipment in the domain 1 so u dont use copression(g711). did you try to put the intipb to a different domain and play with codecs?
the sip phone what codec uses?

[MrAnMo]

If the SIP phone is dialing out. Is that call going through the gd -> pra or through the int-ip -> pra
Could it be that the gateway of the int-ip card don´t have a route to the ip subnetwork from the sip phones?

[jpissot]

Ths sip phones, the GD cards, the INTIP cards and CS are on the same LAN.

[tot3nkopf]

Ths sip phones, the GD cards, the INTIP cards and CS are on the same LAN.

One way voice is a clear indicator of a network problem (usually routing of the RTP stream; but can be a firewall also).

[jpissot]

I have tested at my lab. I use only a switch and I have also the problem.

[jpissot]

If someone have an idea....

Why I have no audio with my cristal mediagateway???

Thank you.

[alex]

If someone have an idea....

Use port mirroring on a switch and take wireshark trace of such a call.
Check which addresses is used for RTP and are they really reachable by both parties.

[jpissot]

I have already made traces.

Communication with UA MG : GD @ and SIP Phone @ . 2 RTP flux
Communication with UA cristal INTIP @ and SIP phone @ but no reverse rtp flux.

Adresses are reachable by both parties.

Includes wireshark traces.


Thank you for your help!!!

[jpissot]

Wireshark Traces can be downloaded at this link :

http://dl.free.fr/gs1D5Edht