Hi,
Hoping someone can help with a Jabber MRA problem. This seems to be affecting mainly IOS (TCT) and Android (BOT) devices and I reckon it's codec related. The problem is we cannot dial any external landline or cell \ mobile numbers through MRA; this happens when the MRA device is connected to WiFi and 4G\cell. The Called party device rings once and then the call automatically disconnects. The call logs report "501 Not Implemented" and "Reason: Q.850;cause=65".
Call route is Jabber MRA device > Expressway-E > Expressway-C > CUCM > SIP Trunk to SIP provider (BT)
Windows CSF devices work fine through MRA using Jabber v11.7.0.
Calls to internal numbers from any MRA device are OK (on Wifi and 4G).
CUCM is v11.5.1.13900-52
Expressway Edge and Core are vX8.10.1
Jabber on IOS and Android is latest v11.9.1
I've had a look at the call logs and I can see that the INVITE from the IOS \ Android devices includes an additional audio codec (number "111") compared to the CSF:
INVITE from Expressway to CUCM
==========================
v=0
o=tandberg 0 1 IN IP4 xxxxx
s=-
c=IN IP4 xxxxx
b=AS:1024
t=0 0
a=cisco-mari:v1
a=cisco-mari-rate
m=audio 55360 RTP/AVP 114 9 104 105 0 8 18 111 101
a=rtpmap:114 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=24000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 x-ulpfecuc/8000
a=fmtp:111 max_esel=1420;m=8;max_n=32;FEC_ORDER=FEC_SRTP
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=extmap:14/sendrecv http://protocols.cisco.com/timestamp#100us
a=sendrecv
a=rtcp:55361 IN IP4 10.50.100.26
INVITE from CUCM to SIP Trunk - includes the same 111 codec:
=================================================
v=0
o=CiscoSystemsCCM-SIP 4388142 1 IN IP4 xxxxx
s=SIP Call
c=IN IP4 xxxxx
b=TIAS:8000
b=AS:8
t=0 0
a=cisco-mari:v1
a=cisco-mari-rate
m=audio 55360 RTP/AVP 18 114 111 101
a=extmap:14/sendrecv http://protocols.cisco.com/timestamp#100us
a=rtpmap:114 opus/48000/2
a=rtpmap:111 X-ULPFECUC/8000
a=fmtp:111 max_esel=1420;m=8;max_n=32;FEC_ORDER=FEC_SRTP
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:55361 IN IP4 10.50.100.26
OFFER from CIP Trunk back to CUCM
=================================
We see the SIP Trunk offering back the following. The "m=8;max_n=32;FEC_ORDER=FEC_SRTP 0" attribute seems completely out of place.
v=0
o=genband 322521088 1508711079 IN IP4 xxxxx
s=-
c=IN IP4 xxxxx
t=0 0
m=audio 47230 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
m=8;max_n=32;FEC_ORDER=FEC_SRTP 0
a=rtpmap:18 0 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Question then is is this an issue with the Jabber app, CUCM, Expressway or the SIP provider, or something else completely?
Thanks
Lee
