[SIPForum-discussion] Loop detected

Mon Nov 25 06:32:44 UTC 2013

HI keyur Amin,

    Thanks for reply.

If i use ideasip server i did not find any loop detected error this also do
not have BRANCH value in VIA header.please ,check the below logs.If I am
missing any thing please guide me.

Sending request: INVITE sip:8355 at ideasip.com SIP/2.0
From: 8355 <sip:8355 at 192.168.2.232>;tag=2448670443
Via: SIP/2.0/UDP 192.168.2.232:40612
Max-Forwards: 70
To: sip:8355 at ideasip.com
Contact: sip:8355 at 192.168.2.232:40612
Call-ID: 734643002 at 192.168.2.232
CSeq: 1 INVITE
Content-Type: application/sdp
User-Agent: playSIP (LIVE555 Streaming Media v2013.10.25)
Content-Length: 117

v=0
o=- 734643002 0 IN IP4 192.168.2.232
s=playSIP session
c=IN IP4 192.168.2.232
t=0 0
m=audio 8000 RTP/AVP 0

RETRANSMISSION 1, after 0.500000 additional seconds
Sending request: INVITE sip:8355 at ideasip.com SIP/2.0
From: 8355 <sip:8355 at 192.168.2.232>;tag=2448670443
Via: SIP/2.0/UDP 192.168.2.232:40612
Max-Forwards: 70
To: sip:8355 at ideasip.com
Contact: sip:8355 at 192.168.2.232:40612
Call-ID: 734643002 at 192.168.2.232
CSeq: 1 INVITE
Content-Type: application/sdp
User-Agent: playSIP (LIVE555 Streaming Media v2013.10.25)
Content-Length: 117

v=0
o=- 734643002 0 IN IP4 192.168.2.232
s=playSIP session
c=IN IP4 192.168.2.232
t=0 0
m=audio 8000 RTP/AVP 0

RETRANSMISSION 2, after 1.000000 additional seconds
Sending request: INVITE sip:8355 at ideasip.com SIP/2.0
From: 8355 <sip:8355 at 192.168.2.232>;tag=2448670443
Via: SIP/2.0/UDP 192.168.2.232:40612
Max-Forwards: 70
To: sip:8355 at ideasip.com
Contact: sip:8355 at 192.168.2.232:40612
Call-ID: 734643002 at 192.168.2.232
CSeq: 1 INVITE
Content-Type: application/sdp
User-Agent: playSIP (LIVE555 Streaming Media v2013.10.25)
Content-Length: 117

v=0
o=- 734643002 0 IN IP4 192.168.2.232
s=playSIP session
c=IN IP4 192.168.2.232
t=0 0
m=audio 8000 RTP/AVP 0

Received INVITE response: SIP/2.0 100 trying -- your call is important to us
From: 8355 <sip:8355 at 192.168.2.232>;tag=2448670443
Via: SIP/2.0/UDP 192.168.2.232:40612;received=27.34.241.138
To: sip:8355 at ideasip.com
Call-ID: 734643002 at 192.168.2.232
CSeq: 1 INVITE
Server: Sip EXpress router (0.9.6 (i386/linux))
Content-Length: 0
Warning: 392 208.97.25.11:5060 "Noisy feedback tells:  pid=18148
req_src_ip=27.34.241.138 req_src_port=40612 in_uri=sip:8355 at ideasip.comout_uri=
sip:9797918005558355 at 208.97.25.12:5090 via_cnt==1"

Received INVITE response: SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.232:40612;received=27.34.241.138
Record-Route: <sip:208.97.25.11;ftag=2448670443;lr=on>
From: 8355 <sip:8355 at 192.168.2.232>;tag=2448670443
To: sip:8355 at ideasip.com;tag=as4765d58a
Call-ID: 734643002 at 192.168.2.232
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:9797918005558355 at 208.97.25.12:5090>
Content-Type: application/sdp
Content-Length: 184

v=0
o=root 25371 25371 IN IP4 208.97.25.12
s=session
c=IN IP4 208.97.25.12
t=0 0
m=audio 13754 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Opened URL "sip:8355 at ideasip.com", returning a SDP description:
v=0
o=- 734643002 0 IN IP4 192.168.2.232
s=playSIP session
c=IN IP4 192.168.2.232
t=0 0
m=audio 8000 RTP/AVP 0

Created receiver for "audio/PCMU" subsession (client ports 8000-8001)
Setup "audio/PCMU" subsession (client ports 8000-8001)
Created output file: "audio-PCMU-1"
Sending request: ACK sip:8355 at ideasip.com SIP/2.0
From: 8355 <sip:8355 at 192.168.2.232>;tag=2448670443
Via: SIP/2.0/UDP 192.168.2.232:40612
Max-Forwards: 70
To: sip:8355 at ideasip.com;tag=as4765d58a
Call-ID: 734643002 at 192.168.2.232
CSeq: 1 ACK
Content-Length: 0

Started playing session
Receiving streamed data (signal with "kill -HUP 16507" or "kill -USR1
16507" to terminate)...

Thanks,
 Raghav.

On Sat, Nov 23, 2013 at 12:24 AM, Keyur Amin <fossil0681 at yahoo.com> wrote:

> You do not have any branch value in Via header.
>
> *Thanks,*
> *Keyur Amin*
> ------------------------------
>
>
>
>
>   On Wednesday, November 20, 2013 12:05 PM, Raghav Goud <
> raghavgoud.g at gmail.com> wrote:
>  Hi all,
>   I am newbie to SIP protocol.
>   Here are the Steps i followed to understand call flow of SIP
> 1.  I Registered an SIP ID with iptel.org.
> 2. I downloaded live555 source code which having sip client.
>   when try to communicate with my SIP ID using live555 sip client am
> getting
>  LOOP DETECTED error.
>
>  Here is log of call flow :
> Sending request: INVITE sip:raghav.goud12 at iptel.org SIP/2.0
> From: raghav.goud12 <sip:raghav.goud12 at 192.168.2.232>;tag=612715021
> Via: SIP/2.0/UDP 192.168.2.232:35728
> Max-Forwards: 70
> To: sip:raghav.goud12 at iptel.org
> Contact: sip:raghav.goud12 at 192.168.2.232:35728
> Call-ID: 2041292707 at 192.168.2.232
> CSeq: 1 INVITE
> Content-Type: application/sdp
> User-Agent: playSIP (LIVE555 Streaming Media v2013.10.25)
> Content-Length: 118
>
> v=0
> o=- 2041292707 0 IN IP4 192.168.2.232
> s=playSIP session
> c=IN IP4 192.168.2.232
> t=0 0
> m=audio 8000 RTP/AVP 0
>
> Received INVITE response: SIP/2.0* 482 Loop Detected*
> From: raghav.goud12 <sip:raghav.goud12 at 192.168.2.232>;tag=612715021
> Via: SIP/2.0/UDP 192.168.2.232:35728;received=27.34.241.138
> To: sip:raghav.goud12 at iptel.org;tag=08EEDDD8-528B2827000D7339-AFDFD700
> Call-ID: 2041292707 at 192.168.2.232
> CSeq: 1 INVITE
> Content-Length: 0
>
>
> Failed to get a SDP description for the URL "sip:raghav.goud12 at iptel.org":
> (NULL)
>
> Can some body Tell me what is error here?
>
> Thanks,
> Raghav
>
>
>
>
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