[SR-Users] about Contact Header Junks

Thu Jan 7 08:41:32 CET 2021

Hello,

that can happen when you do many operations over the contact header
(e.g., functions like fix_natted_contact()), see more details for a
similar case at:

  *
https://www.kamailio.org/wiki/tutorials/faq/main#why_parts_of_fromto_header_app

If you need many operations, then you have to use msg_apply_changes() in
between.

Cheers,
Daniel

On 06.01.21 17:05, Devang Dhandhalya wrote:
> in my requirement call coming from sbc to kamailio proxy and we route
> to pbx server in that when 
> kamailio proxy get the call at that time i am removing Contact header
> using remove_hf("Contact") and adding contact Header using
> append_hf("$uac_req(hdrs)") for route to PBX server . in that
> signaling i am getting modified Contact header but old contact header
> not removed properly some junks are remain in the signaling 
>
> scenario : 
>
> SBC ---------> Kamailio proxy ------> PBX
>
> before using remove_hf("Contact") 
>
> INVITE sip:user2@ <mailto:sip%3A5678 at opensips.org>domain.org
> <http://domain.org>;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP
> 10.212.xxx.xxx:port;branch=z9hG4bK-524287-1---8c5ac8642f321a9e
> Max-Forwards: 70
> Contact: <sip:user1 at 10.212.xxx.xxx:39931;transport=UDP>
> To: <sip:user2@ <mailto:sip%3A5678 at opensips.org>domain.org
> <http://domain.org>;transport=UDP>
> From: <sip:user1@ <mailto:sip%3A1234 at opensips.org>domain.org
> <http://domain.org>;transport=UDP>;tag=891fd646
> Call-ID: t9Ff-fCrin_VN1IO9Xf1mA..
> CSeq: 2 INVITE
> Content-Type: application/sdp
> Proxy-Authorization: Digest username="user1",realm="domain.org
> <http://opensips.org>",nonce="5ff5a660afc72502212b4a611bb0d689efadafab",uri="sip:user2 at domain.org
> <mailto:sip%3A5678 at opensips.org>;transport=UDP",response="f926e3742cbdcb8c68ea9b5
> 82ac2dc",cnonce="9b3276d6d37aaa6107a835df8e5b3a87",nc=00000001,qop=auth,algorithm=MD5
> User-Agent: Z 3.15.40006 rv2.8.20
> Allow-Events: presence, kpml, talk
> Content-Length: 243
>
> v=0
> o=Z 0 0 IN IP4 10.212.xxx.xxx
> s=Z
> c=IN IP4 10.212.xxx.xxx
> t=0 0
> m=audio 8000 RTP/AVP 3 110 8 0 97 101
> a=rtpmap:110 speex/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=sendrecv
>
>
> after using remove-hf("Contact") function removing contact header but
> some junks are left and that eats next To header line and breaks
> Invite packet 
>
> INVITE sip:user2 at domain.org:;transport=UDP SIP/2.0
> Record-Route:
> <sip:172.xx.xx.xxx;lr=on;ftag=891fd646;vst=AAAAAFFTQVkuCENVXUEpHwNLAQEOSwdcDhwUcG9ydD1VRFA-;vsf=AAAAAFVCQ1U0ChwlAQMMHgBHHwFJVAYVYW5zcG9ydD1VRFA->
> Via: SIP/2.0/UDP
> 172.xx.xx.xxx;branch=z9hG4bKbe4c.9743b885ad3452287530994a2cad6e50.0
> Via: SIP/2.0/UDP
> 10.212.xxx.xxx:39931;rport=39931;received=172.xx.xx.x;branch=z9hG4bK-524287-1---8c5ac8642f321a9e
> Max-Forwards: 69
> sip:user1 at 172.xx.xx.x:39931;transport=UDPTo: <sip:user2 at domain.org
> <mailto:sip%3Adevang3032 at opensips.org>>
> From: <sip:user1 at domain.org
> <mailto:sip%3Adppatel at opensips.org>>;tag=891fd646
> Call-ID: t9Ff-fCrin_VN1IO9Xf1mA..
> CSeq: 2 INVITE
> Content-Type: application/sdp
> User-Agent: Z 3.15.40006 rv2.8.20
> Allow-Events: presence, kpml, talk
> Content-Length: 534
> Contact: <sip:user2 at 172.xx.xx.xxx:5060
> <http://sip:dppatel@172.16.16.163:5060>>
>
> v=0
> o=Z 0 0 IN IP4 10.212.xxx.xxx
> s=Z
> c=IN IP4 172.xx.xx.xxx
> t=0 0
> m=audio 11210 RTP/AVP 3 110 8 0 97 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:110 speex/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:97 iLBC/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:97 mode=30
> a=fmtp:101 0-16
> a=sendrecv
> a=rtcp:11211
> a=ice-ufrag:lzjregPe
> a=ice-pwd:RtA9x3jUWk4yNpJlzPL60nsRck
> a=candidate:LhtmUSbPt9BJs4ZC 1 UDP 2130706431 172.16.16.163 11210 typ host
> a=candidate:LhtmUSbPt9BJs4ZC 2 UDP 2130706430 172.16.16.163 11211 typ host
>
>
> As you can see Contact header is gone but
> sip:1234 at 172.xx.xx.x:39931;transport=UDP part left before To header .
> and its breake invite packet .
>
>
> Any suggestion will be highly appreciated.
>
>
>
>
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-- 
Daniel-Constantin Mierla -- www.asipto.com
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