On Thu, 2009-07-02 at 12:59 -0400, Matt Keys wrote: > > I'm testing a SIP trunk to Covista Communications. I'm able to receive > and send calls but there's two catches: > > 1. Placing a call from internal to my cell phone, I do not hear ring > tones but the cell phone will ring and caller ID is shown properly. If > I answer the cell I do not hear any audio in either direction however > according to captures and the phones the call is connected. > > 2. Inbound calls (cell phone to internal) I hear a ring tone and also > see correct caller ID but there's still no audio after answering. > Again, all appears normal on captures and the phones. > > RTP captures are showing a difference in codecs. G.711 on outbound, > example: > > 22.588943 172.16.1.200 -> 172.16.1.2 RTP PT=ITU-T G.711 PCMU, > SSRC=0x94955FF, Seq=27757, Time=688410696 > > and G.729 inbound, example: > > 6.758089 172.16.1.200 -> 172.16.1.2 RTP PT=ITU-T G.729, > SSRC=0x3CA4274A, Seq=25643, Time=1257971077 > > Could this be the problem?
There is no rule that says codecs must be the same in both directions. I doubt this is your problem - it is much more likely a NAT traversal issue. Is Covista Communications doing any NAT compensation on their end - if so, see if you can turn it _off_ so that only one end is compensating for the NAT. _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
