RTP Problem

[cangurer]

Hi,

We have RTP ip problem between 3cx and our voip provider.
This issue cause voice issue and users cannot hear customers voice sometimes.
This is not permanent, it happens randomly especially if we have high sim calls around 35-40

Our Voip provider says we see Yealink ip phone's ip instead of 3cx ip and that cause the issue.

here is an example log: first one has a problem but second one is ok.

Any idea?


INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.37.3.86:5060;branch=z9hG4bK-d8754z-4711260f7a0e0a0c-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: "02123450062"<sip:[email protected]:5060>;tag=7e06d947
Call-ID: MmRkMTgxOGM3Y2VkMDk0MmJkZmRkYTRkMTI0OWI4ODg.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 12.0.37098.738 (34969)
Content-Length: 310

v=0
o=- 21617 21617 IN IP4 192.168.10.226
s=SDP data
c=IN IP4 192.168.10.226
t=0 0
m=audio 11780 RTP/AVP 8 0 18 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20



INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.37.3.86:5060;branch=z9hG4bK-d8754z-d622f06f8839df1a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: "02123450062"<sip:[email protected]:5060>;tag=c329c841
Call-ID: ZWY1ZDMxOGIzN2FiMzIzOTA2MTVlODFmZDdjMDk5NDE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 12.0.37098.738 (34969)
Content-Length: 320

v=0
o=3cxPS 450032041984 57596182529 IN IP4 10.37.3.86
s=3cxPS Audio call
c=IN IP4 10.37.3.86
t=0 0
m=audio 9048 RTP/AVP 8 0 18 9 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000

 

[cangurer]

I just realize that, problematic calls uses port from invalid range,

m=audio 11780 RTP/AVP 8 0 18 9 101