Hi, When sending invite between 2 users Asterisk removes ICE candidates from original invite and uses its own external IP. Is there any configuration I can set to avoid this?
SIP stack:
db1 -> Server
INVITE sip:db2@184.73.165.182 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.82:55257;branch=z9hG4bK2131130297;rport
From: sip:db1@184.73.165.182;tag=152566065
To: sip:db2@184.73.165.182
Contact: sip:db1@192.168.1.82:55257;transport=udp;+g.oma.sip-im;language=“en,fr”;+g.3gpp.icsi-ref=“urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: a98d1760-6187-0140-faca-32012473ba0d
CSeq: 1863079351 INVITE
Content-Type: application/sdp
Content-Length: 1505
Max-Forwards: 70
Authorization: Digest username=“db1”,realm=“asterisk”,nonce=“3fb01621”,uri="sip:db2@184.73.165.182”,response=“390fb09c80c215fe375e605f61da5e32”,algorithm=MD5
Route: sip:184.73.165.182:5060;lr;transport=udp
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.0.498 (doubango r715 - GT-I9100)
P-Preferred-Identity: sip:db1@184.73.165.182
Supported: 100rel
v=0
o=doubango 1983 678901 IN IP4 192.168.1.82
s=-
c=IN IP4 192.168.1.82
t=0 0
m=audio 10370 RTP/AVP 3 101
c=IN IP4 192.168.1.82
a=ptime:20
a=rtpmap:3 GSM/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=tcap:1 RTP/AVPF
a=pcfg:1 t=1
a=sendrecv
a=rtcp-mux
a=ssrc:3117699472 cname:ldjWoB60jbyQlR6e
a=ssrc:3117699472 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:3117699472 label:Doubango
a=ice-ufrag:oCt0zgp7aOlYv3T
a=ice-pwd:EWCXvllhngTWApk9rS8hl
a=mid:audio
a=rtcp-mux
a=candidate:00host00 1 UDP 2130706175 192.168.1.82 10370 typ host
a=candidate:00srflx00 1 UDP 1694498815 62.219.142.171 30180 typ srflx
a=curr:qos e2e none
a=des:qos none e2e sendrecv
m=video 33530 RTP/AVP 34 103 104
c=IN IP4 192.168.1.82
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=2;SQCIF=2
a=rtpmap:103 H263-1998/90000
a=fmtp:103 QCIF=2;SQCIF=2
a=rtpmap:104 H264/90000
a=imageattr:104 recv [x=[128:16:320],y=[96:16:240]] send [x=[128:16:320],y=[96:16:240]]
a=fmtp:104 profile-level-id=42000c; packetization-mode=1
a=tcap:1 RTP/AVPF
a=pcfg:1 t=1
a=sendrecv
a=rtcp-mux
a=ssrc:3658188517 cname:ldjWoB60jbyQlR6e
a=ssrc:3658188517 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:3658188517 label:Doubango
a=ice-ufrag:li0TjgtFOnCnI3a
a=ice-pwd:zPYSG9Njtk3TVtOTGALXT
a=mid:video
a=rtcp-mux
a=candidate:00host00 1 UDP 2130706175 192.168.1.82 33530 typ host
a=candidate:00srflx00 1 UDP 1694498815 62.219.142.171 17915 typ srflx
a=curr:qos e2e none
a=des:qos none e2e sendrecv
Server -> db2
INVITE sip:db2@192.168.1.94:53534;transport=udp SIP/2.0
Via: SIP/2.0/UDP 184.73.165.182:5060;branch=z9hG4bK5d975f4c;rport
Max-Forwards: 70
From: “db1” sip:db1@184.73.165.182;tag=as4ef637f6
To: sip:db2@192.168.1.94:53534;transport=udp
Contact: sip:db1@184.73.165.182:5060
Call-ID: 0018461944b91bab7b12a26930050caf@184.73.165.182:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 19 Jul 2012 11:15:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 421
v=0
o=root 1284307528 1284307528 IN IP4 184.73.165.182
s=Asterisk PBX 10.5.1
c=IN IP4 184.73.165.182
b=CT:5000
t=0 0
m=audio 18604 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 19364 RTP/AVP 99 34 98
a=rtpmap:99 H264/90000
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=sendrecv
Thanks,
Gadi