[SR-Users] Audio quality issue
Austin Einter
austin.einter at gmail.com
Sun Oct 9 23:18:33 CEST 2011
Hi All
Thanks for your kind answer.
The call flow looks as below
I have two doubts here
1. My UA is just behind the Modem, and in Kamailio config file I have
enabled WITH_NAT, will this lead to any kind of problem
2. In kamailio proxy I am using force_rtp_proxy and unforce_rtp_proxy
instead of rtpproxy_offer/rtpproxy_answer. Not sure whats the corresponding
api for unforce_rtp_proxy.
will this lead to any issues.
Regards
Austin.
INVITE sip:919731573290 at 134.121.32.130:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:53489
;rport;branch=z9hG4bKPj0052793130024dda88418cf7a392b7ae
Max-Forwards: 70
From: sip:austin at 134.121.32.130;tag=8c2e350c064e417c96bda1378470fd46
To: sip:919731573290 at 134.121.32.130
Contact: <sip:austin at 192.168.1.2:53489;ob>
Call-ID: b637fa62393a45a0a58633c1a8f43a86
CSeq: 14417 INVITE
Route: <sip:134.33.8.138:5060;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: VoIP Client v1.01
Proxy-Authorization: Digest username="austin", realm="VoipSwitch",
nonce="131819433109160428210053141040", uri="
sip:919731573290 at 134.121.32.130:5060",
response="935c3130fe07e2413ccf127d5fb6b9d1"
Content-Type: application/sdp
Content-Length: 271
v=0
o=- 3527202931 3527202931 IN IP4 192.168.1.2
s=pjmedia
c=IN IP4 192.168.1.2
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 18 4 96
a=rtcp:4001 IN IP4 192.168.1.2
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=sendrecv
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
SIP/2.0 100 trying
Via: SIP/2.0/UDP 192.168.1.2:53489
;rport=13341;branch=z9hG4bKPj0052793130024dda88418cf7a392b7ae;received=122.178.237.67
From: sip:austin at 134.121.32.130;tag=8c2e350c064e417c96bda1378470fd46
To: sip:919731573290 at 134.121.32.130
Call-ID: b637fa62393a45a0a58633c1a8f43a86
CSeq: 14417 INVITE
Server: kamailio (3.1.5 (i386/linux))
Content-Length: 0
SIP/2.0 183 Session Progress
CSeq: 14417 INVITE
Via: SIP/2.0/UDP 192.168.1.2:53489
;branch=z9hG4bKPj0052793130024dda88418cf7a392b7ae
From: sip:austin at 134.121.32.130;tag=8c2e350c064e417c96bda1378470fd46
Call-ID: b637fa62393a45a0a58633c1a8f43a86
To: sip:919731573290 at 134.121.32.130;tag=09100511163117092280006157
Contact: <sip:134.121.32.130:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 241
Record-Route: <sip:134.33.8.138;lr=on;nat=yes>
v=0
o=VoipSwitch 6156 7156 IN IP4 134.33.8.138
s=VoipSIP
i=Audio Session
c=IN IP4 134.33.8.138
t=0 0
m=audio 46976 RTP/AVP 18 96
a=rtpmap:18 G729/8000/1
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
a=nortpproxy:yes
SIP/2.0 200 OK
CSeq: 14417 INVITE
Via: SIP/2.0/UDP 192.168.1.2:53489
;branch=z9hG4bKPj0052793130024dda88418cf7a392b7ae
From: sip:austin at 134.121.32.130;tag=8c2e350c064e417c96bda1378470fd46
Call-ID: b637fa62393a45a0a58633c1a8f43a86
To: sip:919731573290 at 134.121.32.130;tag=09100511163117092280006157
Contact: <sip:134.121.32.130:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 241
Record-Route: <sip:134.33.8.138;lr=on;nat=yes>
v=0
o=VoipSwitch 6156 7156 IN IP4 134.33.8.138
s=VoipSIP
i=Audio Session
c=IN IP4 134.33.8.138
t=0 0
m=audio 46976 RTP/AVP 18 96
a=rtpmap:18 G729/8000/1
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
a=nortpproxy:yes
ACK sip:134.121.32.130:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:53489
;rport;branch=z9hG4bKPj73092b1de9aa4d4498adac484efacfda
Max-Forwards: 70
From: sip:austin at 134.121.32.130;tag=8c2e350c064e417c96bda1378470fd46
To: sip:919731573290 at 134.121.32.130;tag=09100511163117092280006157
Call-ID: b637fa62393a45a0a58633c1a8f43a86
CSeq: 14417 ACK
Route: <sip:134.33.8.138;lr;nat=yes>
Content-Length: 0
BYE sip:austin at 122.178.237.67:13341;ob SIP/2.0
Max-Forwards: 10
CSeq: 1 BYE
Via: SIP/2.0/UDP 134.33.8.138;branch=z9hG4bK029.52d62945.0
Via: SIP/2.0/UDP 134.121.32.130:5060
;rport=5060;branch=z9hG4bK091005111656091709252938
From: sip:919731573290 at 134.121.32.130;tag=09100511163117092280006157
Call-ID: b637fa62393a45a0a58633c1a8f43a86
To: sip:austin at 134.121.32.130;tag=8c2e350c064e417c96bda1378470fd46
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP
134.33.8.138;received=134.33.8.138;branch=z9hG4bK029.52d62945.0
Via: SIP/2.0/UDP 134.121.32.130:5060
;rport=5060;branch=z9hG4bK091005111656091709252938
Call-ID: b637fa62393a45a0a58633c1a8f43a86
From: <sip:919731573290 at 134.121.32.130>;tag=09100511163117092280006157
To: <sip:austin at 134.121.32.130>;tag=8c2e350c064e417c96bda1378470fd46
CSeq: 1 BYE
Content-Length: 0
On Sun, Oct 9, 2011 at 11:50 AM, Sammy Govind <govoiper at gmail.com> wrote:
> Hey,
> Can you send in the SIP/SDP invites. I suspect the codecs issue here.
> --
> Regards,
> Sammy
>
>
> On Sun, Oct 9, 2011 at 8:57 AM, Austin Einter <austin.einter at gmail.com>wrote:
>
>> Hi
>> I am using Kamailio 3.1.5 . I am using RTP proxy also.
>> I have used default kamailio.cfg.sample fiile , and just added line
>> #!define WITH_NAT.
>>
>> I have another Main proxy. I wanted all my signalling and media packets
>> should just pass through machine where Kamailio and RTP proxy are running.
>>
>> With this I found, call is established, all signalling and media packets
>> are passing through kamailio / rtp-proxy.
>> So far so good.
>>
>> One way audio stream (from called party to calling party) quality is good.
>> The other audio stream (from calling party to called party is very bad.
>>
>> Did anybody face this issue? Please help me to sort out this issue audio
>> quality issue.
>>
>> Regards
>> Austin
>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
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>
>
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