Hello; i think i found problem. it is textops module.Appending new header doesnt increase content length. in addition , remove_hf doesnt work correctly that doesnt remove headers.
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Hello,
On 08/04/16 08:06, ycaner wrote:
Hello; i think i found problem. it is textops module.Appending new header doesnt increase content length. in addition , remove_hf doesnt work correctly that doesnt remove headers.
the value of content lenght is about the body size, it is not affected by new headers or removing old headers.
What do you mean by remove_hf() is not working? How you try it? Can you send here how you use remove_hf() in config and the incoming, plus the outgoing sip message?
Cheers, Daniel
Hello Daniel;
in request route when invite comes, a header is added like P-SrcUserid. when i added new header , it truncate sdpbody. before t_relay(), i try to remove header, and it didnt. main problem is why truncates sdpbody when i added new header?
Thanks.
Works fine sip messages: http://pastebin.com/t7v9c2vW removes coming headers cfg,but i couldnt remove P-Provider and P-Uniqueid : http://pastebin.com/9tB7Lx3U
truncateting sip message : http://pastebin.com/DNuuJxhM
removes coming headers cfg,but i couldnt remove P-Provider and P-Uniqueid and added P-SrcUserid : http://pastebin.com/XxhcV55v
08.04.2016 09:22 tarihinde Daniel-Constantin Mierla-6 [via SIP Router] yazdı:
Hello,
On 08/04/16 08:06, ycaner wrote:
> Hello;
> i think i found problem. it is textops module.Appending new header doesnt
> increase content length. in addition , remove_hf doesnt work correctly that
> doesnt remove headers.
>
the value of content lenght is about the body size, it is not affected
by new headers or removing old headers.
What do you mean by remove_hf() is not working? How you try it? Can you
send here how you use remove_hf() in config and the incoming, plus the
outgoing sip message?
Cheers,
Daniel
-- Daniel-Constantin Mierla
http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
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Hello,
how did you asserted the fact that the message is truncated? Is the next SIP hop not retrieving it with full content? Or just you looked at some traces on the net, if this one, note that some network sniffing tools are having their limitations in what they display.
Daniel
On 08/04/16 09:30, ycaner wrote:
Hello Daniel;
in request route when invite comes, a header is added likeP-SrcUserid. when i added new header , it truncate sdpbody. before t_relay(), i try to remove header, and it didnt. main problem is why truncates sdpbody when i added new header?
Thanks.
Works fine sip messages: http://pastebin.com/t7v9c2vW removes coming headers cfg,but i couldnt remove P-Provider and P-Uniqueid :http://pastebin.com/9tB7Lx3U
truncateting sip message : http://pastebin.com/DNuuJxhM
removes coming headers cfg,but i couldnt remove P-Provider and P-Uniqueid and added P-SrcUserid : http://pastebin.com/XxhcV55v
08.04.2016 09:22 tarihinde Daniel-Constantin Mierla-6 [via SIP Router] yazdı:
Hello,
On 08/04/16 08:06, ycaner wrote:
Hello; i think i found problem. it is textops module.Appending new header
doesnt
increase content length. in addition , remove_hf doesnt work
correctly that
doesnt remove headers.
the value of content lenght is about the body size, it is not affected by new headers or removing old headers.
What do you mean by remove_hf() is not working? How you try it? Can you send here how you use remove_hf() in config and the incoming, plus the outgoing sip message?
Cheers, Daniel
-- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda http://twitter.com/#%21/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
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Hello Daniel; source and destination is different machines and ips. they are sniffed by tshark and they are the same and truncated. i tried to remove all extra addes , include Record-Route and send messsage , no truncate. and then started to add headers and BOOM. after 5 headers , it is truncated. configuration become simple to testing one by one. i couldnt find way to solve it. i tried to sniff with tcpdump and sdp body result here , truncated.;
Call Flows ---
UAC1 --> Kamailio1--->kamailio2-->UAC2 --truncated sdp body
UAC1 --> Kamailio2--->UAC2 --No problem
Destination ip tcpdump:
host-kamailio2ip.sip-tls > uac2ip.sip: SIP, length: 1472 INVITE sip:1021uac2num@uac2ip:5060 SIP/2.0 Via: SIP/2.0/UDP kamailio2ip:5061;branch=z9hG4bK61c7.f5edc3dbd9effe58f3d04a6b77b11f13.0 Via: SIP/2.0/UDP kamailio1:5061;rport=5061;branch=z9hG4bK61c7.189d12e4e39ec53abb760692e8a8afee.0 Via: SIP/2.0/UDP 192.168.0.225:10251;received=uac1ip;branch=z9hG4bK702835784163333;rport=17762 From: uac1 <sip:uac1@kamailio1:5061>;tag=1848614912 To: "0uac2num" <sip:0uac2num@kamailio1;user=phone> Call-ID: 228601536217236-500684316713@uac1ip CSeq: 2 INVITE Contact: <sip:uac1@uac1ip:17762> Proxy-Authorization: Digest username="uac1", realm="kamailio1", nonce="Vwd5PlcHeBLSAh0pqSU3AvKzvNAD+QGC", uri="sip:0uac2num@kamailio1;user=phone", response="d296105a65958144cf1b60c921aef11b", algorithm=MD5 Max-Forwards: 68 Supported: replaces, join, path User-Agent: Fanvil C58/C58P 2.2.41.13 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 276 P-Provider:kamailio1 P-Unique:srid-5707759b-29c4-1 atik:1231345123 atik2:1231345123 atik3:1231345123 atik4:1231345123 atik5:1231345123
v=0 o=8503027337 2007313085 1038110647 IN IP4 192.168.0.225 s=A conversation c=IN IP4 192.168.0.225 t=0 0 m=audio 10208 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:1[|sip]
08.04.2016 12:03 tarihinde Daniel-Constantin Mierla-6 [via SIP Router] yazdı:
Hello,
how did you asserted the fact that the message is truncated? Is the next SIP hop not retrieving it with full content? Or just you looked at some traces on the net, if this one, note that some network sniffing tools are having their limitations in what they display.
Daniel
On 08/04/16 09:30, ycaner wrote:
Hello Daniel;
in request route when invite comes, a header is added like P-SrcUserid. when i added new header , it truncate sdpbody. before t_relay(), i try to remove header, and it didnt. main problem is why truncates sdpbody when i added new header?
Thanks.
Works fine sip messages: http://pastebin.com/t7v9c2vW removes coming headers cfg,but i couldnt remove P-Provider
and P-Uniqueid : http://pastebin.com/9tB7Lx3U
truncateting sip message : http://pastebin.com/DNuuJxhM
removes coming headers cfg,but i couldnt remove P-Provider and P-Uniqueid and added P-SrcUserid : http://pastebin.com/XxhcV55v
08.04.2016 09:22 tarihinde Daniel-Constantin Mierla-6 [via SIP Router] yazdı:
Hello,
On 08/04/16 08:06, ycaner wrote: > Hello; > i think i found problem. it is textops module.Appending new header doesnt > increase content length. in addition , remove_hf doesnt work correctly that > doesnt remove headers. > the value of content lenght is about the body size, it is not affected by new headers or removing old headers.
What do you mean by remove_hf() is not working? How you try it? Can you send here how you use remove_hf() in config and the incoming, plus the outgoing sip message?
Cheers, Daniel
-- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
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On 4/8/16 5:27 AM, ycaner wrote:
Hello Daniel; source and destination is different machines and ips. they are sniffed by tshark and they are the same and truncated.
Maybe they are not truncated by rather fragmented. If you are capturing packets by UDP Port you will not see the fragments. Try to capture again by IP only and you will see all packets.
i tried to remove all extra addes , include Record-Route and sendmesssage , no truncate. and then started to add headers and BOOM. after 5 headers , it is truncated. configuration become simple to testing one by one. i couldnt find way to solve it. i tried to sniff with tcpdump and sdp body result here , truncated.;
Call Flows ---
UAC1 --> Kamailio1--->kamailio2-->UAC2 --truncated sdp body
UAC1 --> Kamailio2--->UAC2 --No problem
Destination ip tcpdump:
host-kamailio2ip.sip-tls > uac2ip.sip: SIP, length: 1472 INVITE sip:1021uac2num@uac2ip:5060 SIP/2.0 Via: SIP/2.0/UDP kamailio2ip:5061;branch=z9hG4bK61c7.f5edc3dbd9effe58f3d04a6b77b11f13.0 Via: SIP/2.0/UDP kamailio1:5061;rport=5061;branch=z9hG4bK61c7.189d12e4e39ec53abb760692e8a8afee.0 Via: SIP/2.0/UDP 192.168.0.225:10251;received=uac1ip;branch=z9hG4bK702835784163333;rport=17762 From: uac1 sip:uac1@kamailio1:5061;tag=1848614912 To: "0uac2num" sip:0uac2num@kamailio1;user=phone Call-ID: 228601536217236-500684316713@uac1ip CSeq: 2 INVITE Contact: sip:uac1@uac1ip:17762 Proxy-Authorization: Digest username="uac1", realm="kamailio1", nonce="Vwd5PlcHeBLSAh0pqSU3AvKzvNAD+QGC", uri="sip:0uac2num@kamailio1;user=phone", response="d296105a65958144cf1b60c921aef11b", algorithm=MD5 Max-Forwards: 68 Supported: replaces, join, path User-Agent: Fanvil C58/C58P 2.2.41.13 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 276 P-Provider:kamailio1 P-Unique:srid-5707759b-29c4-1 atik:1231345123 atik2:1231345123 atik3:1231345123 atik4:1231345123 atik5:1231345123
v=0 o=8503027337 2007313085 1038110647 IN IP4 192.168.0.225 s=A conversation c=IN IP4 192.168.0.225 t=0 0 m=audio 10208 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:1[|sip]08.04.2016 12:03 tarihinde Daniel-Constantin Mierla-6 [via SIP Router] yazdı:
Hello,
how did you asserted the fact that the message is truncated? Is the next SIP hop not retrieving it with full content? Or just you looked at some traces on the net, if this one, note that some network sniffing tools are having their limitations in what they display.
Daniel
On 08/04/16 09:30, ycaner wrote:
Hello Daniel;
in request route when invite comes, a header is added likeP-SrcUserid. when i added new header , it truncate sdpbody. before t_relay(), i try to remove header, and it didnt. main problem is why truncates sdpbody when i added new header?
Thanks.
Works fine sip messages: http://pastebin.com/t7v9c2vW removes coming headers cfg,but i couldnt remove P-Provider and P-Uniqueid :http://pastebin.com/9tB7Lx3U
truncateting sip message : http://pastebin.com/DNuuJxhM
removes coming headers cfg,but i couldnt remove P-Provider and P-Uniqueid and added P-SrcUserid : http://pastebin.com/XxhcV55v
08.04.2016 09:22 tarihinde Daniel-Constantin Mierla-6 [via SIP Router] yazdı:
Hello,
On 08/04/16 08:06, ycaner wrote:
Hello; i think i found problem. it is textops module.Appending new
header doesnt
increase content length. in addition , remove_hf doesnt work
correctly that
doesnt remove headers.
the value of content lenght is about the body size, it is not affected by new headers or removing old headers.
What do you mean by remove_hf() is not working? How you try it? Can you send here how you use remove_hf() in config and the incoming, plus the outgoing sip message?
Cheers, Daniel
-- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda http://twitter.com/#%21/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
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Hello; I saw it. it is fragmented IP and cant resolve it wireshark. But i cant understand one thing, First Call Flows makes fragmantion but second call flows doesnt fragmantaion even if it is larger than first call flows.
Thanks.
08.04.2016 14:40 tarihinde Andres-27 [via SIP Router] yazdı:
On 4/8/16 5:27 AM, ycaner wrote:
Hello Daniel; source and destination is different machines and ips. they are sniffed by tshark and they are the same and truncated.
Maybe they are not truncated by rather fragmented. If you are capturing packets by UDP Port you will not see the fragments. Try to capture again by IP only and you will see all packets. i tried to remove all extra addes , include Record-Route and send messsage , no truncate. and then started to add headers and BOOM. after 5 headers , it is truncated. configuration become simple to testing one by one. i couldnt find way to solve it. i tried to sniff with tcpdump and sdp body result here , truncated.;
Call Flows ---
UAC1 --> Kamailio1--->kamailio2-->UAC2 --truncated sdp body
UAC1 --> Kamailio2--->UAC2 --No problem
Destination ip tcpdump:
host-kamailio2ip.sip-tls > uac2ip.sip: SIP, length: 1472 INVITE sip:1021uac2num@uac2ip:5060 SIP/2.0 Via: SIP/2.0/UDP kamailio2ip:5061;branch=z9hG4bK61c7.f5edc3dbd9effe58f3d04a6b77b11f13.0 Via: SIP/2.0/UDP kamailio1:5061;rport=5061;branch=z9hG4bK61c7.189d12e4e39ec53abb760692e8a8afee.0 Via: SIP/2.0/UDP 192.168.0.225:10251;received=uac1ip;branch=z9hG4bK702835784163333;rport=17762 From: uac1 <sip:uac1@kamailio1:5061>;tag=1848614912 To: "0uac2num" <sip:0uac2num@kamailio1;user=phone> Call-ID: 228601536217236-500684316713@uac1ip CSeq: 2 INVITE Contact: <sip:uac1@uac1ip:17762> Proxy-Authorization: Digest username="uac1", realm="kamailio1", nonce="Vwd5PlcHeBLSAh0pqSU3AvKzvNAD+QGC", uri="sip:0uac2num@kamailio1;user=phone", response="d296105a65958144cf1b60c921aef11b", algorithm=MD5 Max-Forwards: 68 Supported: replaces, join, path User-Agent: Fanvil C58/C58P 2.2.41.13 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 276 P-Provider:kamailio1 P-Unique:srid-5707759b-29c4-1 atik:1231345123 atik2:1231345123 atik3:1231345123 atik4:1231345123 atik5:1231345123
v=0 o=8503027337 2007313085 1038110647 IN IP4 192.168.0.225 s=A conversation c=IN IP4 192.168.0.225 t=0 0 m=audio 10208 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:1[|sip]
08.04.2016 12:03 tarihinde Daniel-Constantin Mierla-6 [via SIP Router] yazdı:
Hello,
how did you asserted the fact that the message is truncated? Is the next SIP hop not retrieving it with full content? Or just you looked at some traces on the net, if this one, note that some network sniffing tools are having their limitations in what they display.
Daniel
On 08/04/16 09:30, ycaner wrote:
Hello Daniel;
in request route when invite comes, a header is added like P-SrcUserid. when i added new header , it truncate sdpbody. before t_relay(), i try to remove header, and it didnt. main problem is why truncates sdpbody when i added new header?
Thanks.
Works fine sip messages: http://pastebin.com/t7v9c2vW removes coming headers cfg,but i couldnt remove P-Provider and P-Uniqueid : http://pastebin.com/9tB7Lx3U
truncateting sip message : http://pastebin.com/DNuuJxhM
removes coming headers cfg,but i couldnt remove P-Provider and P-Uniqueid and added P-SrcUserid : http://pastebin.com/XxhcV55v
08.04.2016 09:22 tarihinde Daniel-Constantin Mierla-6 [via SIP Router] yazdı:
Hello,
On 08/04/16 08:06, ycaner wrote: > Hello; > i think i found problem. it is textops module.Appending new header doesnt > increase content length. in addition , remove_hf doesnt work correctly that > doesnt remove headers. > the value of content lenght is about the body size, it is not affected by new headers or removing old headers.
What do you mean by remove_hf() is not working? How you try it? Can you send here how you use remove_hf() in config and the incoming, plus the outgoing sip message?
Cheers, Daniel
-- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
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On 4/8/16 8:19 AM, ycaner wrote:
Hello; I saw it. it is fragmented IP and cant resolve it wireshark. But i cant understand one thing, First Call Flows makes fragmantion but second call flows doesnt fragmantaion even if it is larger than first call flows.
Fragmentation will happen at the Operating System level, not at the Application level. If the medium is Ethernet and the entire content including IP headers exceed 1500 bytes then it gets fragmented. Wireshark can clearly show you the packet length for any complete or fragmented packet.
Thanks.
08.04.2016 14:40 tarihinde Andres-27 [via SIP Router] yazdı:
On 4/8/16 5:27 AM, ycaner wrote:
Hello Daniel; source and destination is different machines and ips. they are sniffed by tshark and they are the same and truncated.
Maybe they are not truncated by rather fragmented. If you are capturing packets by UDP Port you will not see the fragments. Try to capture again by IP only and you will see all packets.
i tried to remove all extra addes , include Record-Route andsend messsage , no truncate. and then started to add headers and BOOM. after 5 headers , it is truncated. configuration become simple to testing one by one. i couldnt find way to solve it. i tried to sniff with tcpdump and sdp body result here , truncated.;
Call Flows ---
UAC1 --> Kamailio1--->kamailio2-->UAC2 --truncated sdp body
UAC1 --> Kamailio2--->UAC2 --No problem
Destination ip tcpdump:
host-kamailio2ip.sip-tls > uac2ip.sip: SIP, length: 1472 INVITE sip:1021uac2num@uac2ip:5060 SIP/2.0 Via: SIP/2.0/UDP kamailio2ip:5061;branch=z9hG4bK61c7.f5edc3dbd9effe58f3d04a6b77b11f13.0 Via: SIP/2.0/UDP kamailio1:5061;rport=5061;branch=z9hG4bK61c7.189d12e4e39ec53abb760692e8a8afee.0 Via: SIP/2.0/UDP 192.168.0.225:10251;received=uac1ip;branch=z9hG4bK702835784163333;rport=17762 From: uac1 sip:uac1@kamailio1:5061;tag=1848614912 To: "0uac2num" sip:0uac2num@kamailio1;user=phone Call-ID: 228601536217236-500684316713@uac1ip CSeq: 2 INVITE Contact: sip:uac1@uac1ip:17762 Proxy-Authorization: Digest username="uac1", realm="kamailio1", nonce="Vwd5PlcHeBLSAh0pqSU3AvKzvNAD+QGC", uri="sip:0uac2num@kamailio1;user=phone", response="d296105a65958144cf1b60c921aef11b", algorithm=MD5 Max-Forwards: 68 Supported: replaces, join, path User-Agent: Fanvil C58/C58P 2.2.41.13 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 276 P-Provider:kamailio1 P-Unique:srid-5707759b-29c4-1 atik:1231345123 atik2:1231345123 atik3:1231345123 atik4:1231345123 atik5:1231345123
v=0 o=8503027337 2007313085 1038110647 IN IP4 192.168.0.225 s=A conversation c=IN IP4 192.168.0.225 t=0 0 m=audio 10208 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:1[|sip]08.04.2016 12:03 tarihinde Daniel-Constantin Mierla-6 [via SIP Router] yazdı:
Hello,
how did you asserted the fact that the message is truncated? Is the next SIP hop not retrieving it with full content? Or just you looked at some traces on the net, if this one, note that some network sniffing tools are having their limitations in what they display.
Daniel
On 08/04/16 09:30, ycaner wrote:
Hello Daniel;
in request route when invite comes, a header is added likeP-SrcUserid. when i added new header , it truncate sdpbody. before t_relay(), i try to remove header, and it didnt. main problem is why truncates sdpbody when i added new header?
Thanks.
Works fine sip messages: http://pastebin.com/t7v9c2vW removes coming headers cfg,but i couldnt remove P-Provider and P-Uniqueid :http://pastebin.com/9tB7Lx3U
truncateting sip message : http://pastebin.com/DNuuJxhM
removes coming headers cfg,but i couldnt remove P-Provider and P-Uniqueid and added P-SrcUserid : http://pastebin.com/XxhcV55v
08.04.2016 09:22 tarihinde Daniel-Constantin Mierla-6 [via SIP Router] yazdı:
Hello,
On 08/04/16 08:06, ycaner wrote: > Hello; > i think i found problem. it is textops module.Appending new header doesnt > increase content length. in addition , remove_hf doesnt work correctly that > doesnt remove headers. > the value of content lenght is about the body size, it is not affected by new headers or removing old headers.
What do you mean by remove_hf() is not working? How you try it? Can you send here how you use remove_hf() in config and the incoming, plus the outgoing sip message?
Cheers, Daniel
-- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda http://twitter.com/#%21/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
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After i saw it , i search on net , yes, i got it but one thing make me think wrong because reassabmle fragmented IPv4 datagrams on wireshark was disabled by me ,long time ago so it is why i fell in a block hole.
Thanks for advices
10.04.2016 00:46 tarihinde Andres-27 [via SIP Router] yazdı:
On 4/8/16 8:19 AM, ycaner wrote:
Hello; I saw it. it is fragmented IP and cant resolve it wireshark. But i cant understand one thing, First Call Flows makes fragmantion but second call flows doesnt fragmantaion even if it is larger than first call flows.
Fragmentation will happen at the Operating System level, not at the Application level. If the medium is Ethernet and the entire content including IP headers exceed 1500 bytes then it gets fragmented. Wireshark can clearly show you the packet length for any complete or fragmented packet. Thanks.
08.04.2016 14:40 tarihinde Andres-27 [via SIP Router] yazdı:
On 4/8/16 5:27 AM, ycaner wrote:
Hello Daniel; source and destination is different machines and ips. they are sniffed by tshark and they are the same and truncated.
Maybe they are not truncated by rather fragmented. If you are capturing packets by UDP Port you will not see the fragments. Try to capture again by IP only and you will see all packets. i tried to remove all extra addes , include Record-Route and send messsage , no truncate. and then started to add headers and BOOM. after 5 headers , it is truncated. configuration become simple to testing one by one. i couldnt find way to solve it. i tried to sniff with tcpdump and sdp body result here , truncated.;
Call Flows ---
UAC1 --> Kamailio1--->kamailio2-->UAC2 --truncated sdp body
UAC1 --> Kamailio2--->UAC2 --No problem
Destination ip tcpdump:
host-kamailio2ip.sip-tls > uac2ip.sip: SIP, length: 1472 INVITE sip:1021uac2num@uac2ip:5060 SIP/2.0 Via: SIP/2.0/UDP kamailio2ip:5061;branch=z9hG4bK61c7.f5edc3dbd9effe58f3d04a6b77b11f13.0 Via: SIP/2.0/UDP kamailio1:5061;rport=5061;branch=z9hG4bK61c7.189d12e4e39ec53abb760692e8a8afee.0 Via: SIP/2.0/UDP 192.168.0.225:10251;received=uac1ip;branch=z9hG4bK702835784163333;rport=17762 From: uac1 <sip:uac1@kamailio1:5061>;tag=1848614912 To: "0uac2num" <sip:0uac2num@kamailio1;user=phone> Call-ID: 228601536217236-500684316713@uac1ip CSeq: 2 INVITE Contact: <sip:uac1@uac1ip:17762> Proxy-Authorization: Digest username="uac1", realm="kamailio1", nonce="Vwd5PlcHeBLSAh0pqSU3AvKzvNAD+QGC", uri="sip:0uac2num@kamailio1;user=phone", response="d296105a65958144cf1b60c921aef11b", algorithm=MD5 Max-Forwards: 68 Supported: replaces, join, path User-Agent: Fanvil C58/C58P 2.2.41.13 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 276 P-Provider:kamailio1 P-Unique:srid-5707759b-29c4-1 atik:1231345123 atik2:1231345123 atik3:1231345123 atik4:1231345123 atik5:1231345123
v=0 o=8503027337 2007313085 1038110647 IN IP4 192.168.0.225 s=A conversation c=IN IP4 192.168.0.225 t=0 0 m=audio 10208 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:1[|sip]
08.04.2016 12:03 tarihinde Daniel-Constantin Mierla-6 [via SIP Router] yazdı:
Hello,
how did you asserted the fact that the message is truncated? Is the next SIP hop not retrieving it with full content? Or just you looked at some traces on the net, if this one, note that some network sniffing tools are having their limitations in what they display.
Daniel
On 08/04/16 09:30, ycaner wrote:
Hello Daniel;
in request route when invite comes, a header is added like P-SrcUserid. when i added new header , it truncate sdpbody. before t_relay(), i try to remove header, and it didnt. main problem is why truncates sdpbody when i added new header?
Thanks.
Works fine sip messages: http://pastebin.com/t7v9c2vW removes coming headers cfg,but i couldnt remove P-Provider and P-Uniqueid : http://pastebin.com/9tB7Lx3U
truncateting sip message : http://pastebin.com/DNuuJxhM
removes coming headers cfg,but i couldnt remove P-Provider and P-Uniqueid and added P-SrcUserid : http://pastebin.com/XxhcV55v
08.04.2016 09:22 tarihinde Daniel-Constantin Mierla-6 [via SIP Router] yazdı:
Hello,
On 08/04/16 08:06, ycaner wrote: > Hello; > i think i found problem. it is textops module.Appending new header doesnt > increase content length. in addition , remove_hf doesnt work correctly that > doesnt remove headers. > the value of content lenght is about the body size, it is not affected by new headers or removing old headers.
What do you mean by remove_hf() is not working? How you try it? Can you send here how you use remove_hf() in config and the incoming, plus the outgoing sip message?
Cheers, Daniel
-- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
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Hello Daniel; 3 day ago , i just started to work on rtpproxy. My call flow like UAC1--> Kamailio1---> Kamailio2+RTPPROXY--> UAC2 but it doesnt work because i saw sdp body has missing parameters but rtp flows like UAC1<---->UAC2. And then i removed first kamailio and i saw it worked correctly and there is no mistake on sdpbody. RTP flows like UAC1--->Kamailio2+RTPPROXY---->UAC2 and SIP Flows like UAC1----> Kamailio2+RTTPROXY-->UAC2. i thought it is really weird because how is possible Kamailio1 effect like that. And then i removed nearly all modules and left simple cfg. i realized that appending header effects it. Could you help me about it, please
Thanks
i solved remove_hf function that kamailio removes header field that adds by self , after msg_apply_changes().
i tested UAC1--->Kamailio2---->UAC2 call flow and added 14 header doesnt have any problem.
Thanks.
08.04.2016 12:03 tarihinde Daniel-Constantin Mierla-6 [via SIP Router] yazdı:
Hello,
how did you asserted the fact that the message is truncated? Is the next SIP hop not retrieving it with full content? Or just you looked at some traces on the net, if this one, note that some network sniffing tools are having their limitations in what they display.
Daniel
On 08/04/16 09:30, ycaner wrote:
Hello Daniel;
in request route when invite comes, a header is added like P-SrcUserid. when i added new header , it truncate sdpbody. before t_relay(), i try to remove header, and it didnt. main problem is why truncates sdpbody when i added new header?
Thanks.
Works fine sip messages: http://pastebin.com/t7v9c2vW removes coming headers cfg,but i couldnt remove P-Provider
and P-Uniqueid : http://pastebin.com/9tB7Lx3U
truncateting sip message : http://pastebin.com/DNuuJxhM
removes coming headers cfg,but i couldnt remove P-Provider and P-Uniqueid and added P-SrcUserid : http://pastebin.com/XxhcV55v
08.04.2016 09:22 tarihinde Daniel-Constantin Mierla-6 [via SIP Router] yazdı:
Hello,
On 08/04/16 08:06, ycaner wrote: > Hello; > i think i found problem. it is textops module.Appending new header doesnt > increase content length. in addition , remove_hf doesnt work correctly that > doesnt remove headers. > the value of content lenght is about the body size, it is not affected by new headers or removing old headers.
What do you mean by remove_hf() is not working? How you try it? Can you send here how you use remove_hf() in config and the incoming, plus the outgoing sip message?
Cheers, Daniel
-- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
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Andres -
Daniel-Constantin Mierla -
Yasin CANER -
ycaner