[Freeswitch-users] No audio on caller side when both side support speex/8000 only

[Tzury Bar Yochay]

Hi,

Owe to the network bandwidth limitations (running on cellular phones
ip link) we are using speex/8000 as our voice codec.

However, when both parties are using that codec the sound is not to be
heard on the caller side.

looking at the log dumps one can see that

a) at the caller side, it supports speex/8000 in pt=102 and receives
from the server speex/8000 in pt=102
b) at the callee side FreeSwitch supports support speex/8000 in pt=98
although it receives from the client speex/8000 in pt=102

When the voice starts caller sends RTP with pt=102 and expect to
receive RTP with pt=102, while the callee sends RTP with pt=98 and
expect to receive RTP with pt=102.

The RTP packets that received in the caller side are with pt=98
instead of 102 and thusly the client drops them.

############### LOG DUMPS ###############

###############  CALLER SIDE

## start msg (TX)
INVITE sip:1002 at SERVER_DOMAIN SIP/2.0
Via: SIP/2.0/UDP
X.X.X.X:64680;rport;branch=z9hG4bKPj3caad40720064a8f124c21cf99b8b1c1
Max-Forwards: 70
From: sip:1001 at SERVER_DOMAIN;tag=6d6ef114e663e48226f2b1e598313a2e
To: sip:1002 at SERVER_DOMAIN
Contact: <sip:1001 at X.X.X.X:64680>
Call-ID: 1473e9e828658e3fb0370fabf2ce8986
CSeq: 23264 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length:   264

v=0
o=- 3461521040 3461521040 IN IP4 X.X.X.X
s=pjmedia
c=IN IP4 X.X.X.X
t=0 0
a=X-nat:8
m=audio 64976 RTP/AVP 102 101
a=rtcp:64980 IN IP4 X.X.X.X
a=rtpmap:102 speex/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--end msg--


## start msg (RX)
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.171.9.67:5060;rport=64680;branch=z9hG4bKPje8deea3c8476457603fe0cc301731002
From: <sip:1001 at SERVER_DOMAIN>;tag=6d6ef114e663e48226f2b1e598313a2e
To: <sip:1002 at SERVER_DOMAIN>;tag=06ym4113FFcHQ
Call-ID: 1473e9e828658e3fb0370fabf2ce8986
CSeq: 23265 INVITE
Contact: <sip:mod_sofia at X.X.X.X:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Require: timer
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Session-Expires: 1800;refresher=uac
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 268

v=0
o=FreeSWITCH 4446028933093139022 3405868075899860026 IN IP4 X.X.X.X
s=FreeSWITCH
c=IN IP4 X.X.X.X
t=0 0
m=audio 26664 RTP/AVP 102 101
a=rtpmap:102 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

--end msg--


###############  CALLEE SIDE


## start msg (RX)
INVITE sip:1002 at X.X.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.X;rport;branch=z9hG4bKgvD702De7e0Se
Max-Forwards: 69
From: "Extension 1001" <sip:1001 at X.X.X.X>;tag=2rH67Q3aa1rpe
To: <sip:1002 at X.X.X.X:5060>
Call-ID: e56c2918-17ad-122d-de9e-40402384297d
CSeq: 120120747 INVITE
Contact: <sip:mod_sofia at X.X.X.X:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 415
Remote-Party-ID: "Extension 1001" <sip:1001 at X.X.X.X>;screen=yes;privacy=off

v=0
o=FreeSWITCH 4131815555116427886 953315150658749217 IN IP4 X.X.X.X
s=FreeSWITCH
c=IN IP4 X.X.X.X
t=0 0
m=audio 26662 RTP/AVP 98 0 8 3 9 99 103 101 13
a=rtpmap:98 SPEEX/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:99 SPEEX/16000
a=rtpmap:103 SPEEX/32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20

--end msg--


## start msg (TX)
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.X;rport=5060;received=X.X.X.X;branch=z9hG4bKgvD702De7e0Se
Call-ID: e56c2918-17ad-122d-de9e-40402384297d
From: "Extension 1001" <sip:1001 at X.X.X.X>;tag=2rH67Q3aa1rpe
To: <sip:1002 at X.X.X.X>;tag=e4f9fb648edef1c48dbc8b8b474409e6
CSeq: 120120747 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Contact: <sip:X.X.X.X:5060>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length:   256

v=0
o=- 3461503025 3461503026 IN IP4 X.X.X.X
s=pjmedia
c=IN IP4 X.X.X.X
t=0 0
a=X-nat:5
m=audio 4000 RTP/AVP 102 101
a=rtcp:4001 IN IP4 X.X.X.X
a=rtpmap:102 speex/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--end msg--


Attached are the 2 files recorded from a call between 2 pjsip clients
that support only speex/8000 codec.

un_FSCallerSide-speexClient.TXT – is the caller side SIP messages.
un_FSAnswerSide-speexClient.TXT – is the answer side of SIP messages.


Is there anything can be done at the configuration level to avoid this?
Thanks in advance for your help

/tzury