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.
Via: SIP/2.0/UDP 192.168.1.2:53489;rport;branch=z9hG4bKPj0052793130024dda88418cf7a392b7ae
Max-Forwards: 70
Contact: <sip:austin@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@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
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
Call-ID: b637fa62393a45a0a58633c1a8f43a86
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
Call-ID: b637fa62393a45a0a58633c1a8f43a86
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
Call-ID: b637fa62393a45a0a58633c1a8f43a86
CSeq: 14417 ACK
Route: <sip:134.33.8.138;lr;nat=yes>
Content-Length: 0
BYE sip:austin@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
Call-ID: b637fa62393a45a0a58633c1a8f43a86
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
CSeq: 1 BYE
Content-Length: 0
On Sun, Oct 9, 2011 at 11:50 AM, Sammy Govind
<govoiper@gmail.com> wrote:
Hey,
Can you send in the SIP/SDP invites. I suspect the codecs issue here.
--
Regards,
Sammy
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
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